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Callweaver и факсы

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uni_advisor
Сообщений: 54

Callweaver и факсы

Решил я протестить callweaver форк 1.2 астериска. Как разработчики заявляют что мол там нормальная поддержка Т38. Собирал стабильную версию и callwea~1.2.0.1, spandsp spandsp-0.0.5pre3. Схема подключения следующая. PSTN -> E1 -> Audiocodes Mediant-1000 -> callweaver -> Шлюз Audiocode MP -> Факсовый аппарат.

Сначала поставил дебаг на медиант. Потом второй раз поставил дебаг на сам шлюз с факсом.
t38udptl=yes в sip.conf, canreinvite=yes, если ставить No тогда не слышно голоса.

В extensions.conf дозвон на шлюз
exten => 4956273690,1,Dial(SIP/${EXTEN})
exten => 4956273690,n,Hangup

Факсы не ходят. Очень печально. На сегодняшний момент более менее заставить работать факсы получилось только с патчами на астер 1.4 от пользователся Cache.
2009-04-21 11:38

uni_advisor
Сообщений: 54

Re: Callweaver и факсы

Привожу дебаги сиповые снятые для Медианта
centos1*CLI> sip debug peer Mediant
SIP Debugging Enabled for IP: 217.28.210.18:5060
12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
OPTIONS sip:217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK35b04882;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as0076a266
To: <sip:217.28.210.18>
Contact: <sip:callweaver@172.16.10.200>
Call-ID: 567b685062c80bcd33991298730527d5@172.16.10.200
CSeq: 102 OPTIONS
User-Agent: CallWeaver
Max-Forwards: 70
Date: Tue, 21 Apr 2009 06:46:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK35b04882;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as0076a266
To: <sip:217.28.210.18>;tag=1c913669632
Call-ID: 567b685062c80bcd33991298730527d5@172.16.10.200
CSeq: 102 OPTIONS
Contact: <sip:217.28.210.18>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
X-Resources: telchs=21/9;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 569

v=0
o=AudiocodesGW 913679469 913679147 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6000 RTP/AVP 4 0 8 18 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=rtcp:6001 IN IP4 217.28.210.18
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 24 lines) ---
Destroying call '567b685062c80bcd33991298730527d5@172.16.10.200'
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
INVITE sip:4956273690@172.16.10.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1165849069
Max-Forwards: 70
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 1 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 572

v=0
o=AudiocodesGW 1165807932 1165807609 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 4 0 8 18 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18
m=image 6342 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 24 lines) ---
Using INVITE request as basis request - 11658383702142009111129@217.28.210.18
Sending to 217.28.210.18 : 5060 (non-NAT)
Found peer 'Mediant'
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Got T.38 offer in SDP
Apr 21 10:46:27 WARNING[3063942032]: channel.c:683 cw_channel_perform_set_t38_status: cw_channel_set_t38_status called with NULL channel at chan_sip.c:4934
Peer audio RTP is at port 217.28.210.18:6340
Found description format G723
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4956273690 in internal (domain 172.16.10.200;user=phone)
list_route: hop: <sip:4957772333@217.28.210.18>
Transmitting (no NAT) to 217.28.210.18:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1165849069;received=217.28.210.18
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4956273690@172.16.10.200>
Content-Length: 0


---
-- Executing [4956273690@internal:1] Dial("SIP/4957772333-0940c8f8", "SIP/4956273690")
-- Called 4956273690
-- SIP/4956273690-7610 is ringing
Transmitting (no NAT) to 217.28.210.18:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1165849069;received=217.28.210.18
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4956273690@172.16.10.200>
Content-Length: 0


---
-- SIP/4956273690-7610 answered SIP/4957772333-0940c8f8
T.38 UDPTL is at port 172.16.10.200:16292...
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1165849069;received=217.28.210.18
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4956273690@172.16.10.200>
Content-Type: application/sdp
Content-Length: 344

v=0
o=root 9046 9046 IN IP4 172.16.10.200
s=session
c=IN IP4 172.16.10.200
t=0 0
m=image 16292 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
ACK sip:4956273690@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1204506600
Max-Forwards: 70
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 1 ACK
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Length: 0


--- (12 headers 0 lines) ---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
We're at 172.16.10.200 port 16292
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK2b174d0d;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9047 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
SIP TIMER: #472: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK2b174d0d;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9047 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK2b174d0d;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 102 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807610 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:4957772333@217.28.210.18>
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK73906ab3;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 102 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK2b174d0d;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 102 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807610 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK38dc416f;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 102 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
T.38 UDPTL is at port 172.16.10.200:16292...
13 headers, 15 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9048 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #474: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9048 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #474: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9048 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #474: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9048 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 295

v=0
o=AudiocodesGW 1165807932 1165807611 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=image 6342 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (13 headers 12 lines) ---
Got T.38 offer in SDP
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK25a011b7;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
We're at 172.16.10.200 port 16292
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4999f5b7;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 104 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9049 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 295

v=0
o=AudiocodesGW 1165807932 1165807611 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=image 6342 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (13 headers 12 lines) ---

<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 295

v=0
o=AudiocodesGW 1165807932 1165807611 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=image 6342 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (13 headers 12 lines) ---

<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK63dca6db;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 103 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 295

v=0
o=AudiocodesGW 1165807932 1165807611 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=image 6342 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (13 headers 12 lines) ---
SIP TIMER: #476: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4999f5b7;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 104 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9049 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4999f5b7;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 104 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807612 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:46:34 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK22a2aa47;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 104 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
We're at 172.16.10.200 port 16292
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1cf53958;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 105 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9050 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4999f5b7;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 104 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807612 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
SIP TIMER: #478: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1cf53958;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 105 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9050 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1cf53958;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 105 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807613 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:46:34 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK26f42054;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 105 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1cf53958;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 105 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807613 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:46:34 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK34fb8030;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 105 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
We're at 172.16.10.200 port 16292
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4b49d1e8;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 106 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9051 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
SIP TIMER: #480: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4b49d1e8;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 106 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9051 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4b49d1e8;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 106 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807614 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4b6125d6;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 106 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4b49d1e8;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 106 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807614 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK0a29de73;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 106 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
INVITE sip:4956273690@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1411969963
Max-Forwards: 70
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 2 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 295

v=0
o=AudiocodesGW 1165807932 1165807615 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=image 6342 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 12 lines) ---
Using INVITE request as basis request - 11658383702142009111129@217.28.210.18
Sending to 217.28.210.18 : 5060 (non-NAT)
Got T.38 offer in SDP
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:46:36 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4956273690-7610 - Call ID 384d57b94da23f131ab43c080cfa69dc@172.16.10.200!
Apr 21 10:46:36 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4956273690-7610 - Call ID 384d57b94da23f131ab43c080cfa69dc@172.16.10.200!
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4956273690-7610 - Call ID 384d57b94da23f131ab43c080cfa69dc@172.16.10.200!
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4956273690-7610 - Call ID 384d57b94da23f131ab43c080cfa69dc@172.16.10.200!
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4956273690-7610 - Call ID 384d57b94da23f131ab43c080cfa69dc@172.16.10.200!
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4956273690-7610 - Call ID 384d57b94da23f131ab43c080cfa69dc@172.16.10.200!
Apr 21 10:46:37 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:46:37 WARNING[3063942032]: chan_sip.c:1564 retrans_pkt: Maximum retries exceeded on transmission 384d57b94da23f131ab43c080cfa69dc@172.16.10.200 for seqno 107 (Non-critical Request)
T.38 UDPTL is at port 172.16.10.200:16292...
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1411969963;received=217.28.210.18
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 2 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4956273690@172.16.10.200>
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9052 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
ACK sip:4956273690@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1419386381
Max-Forwards: 70
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 2 ACK
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Length: 0


--- (12 headers 0 lines) ---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
We're at 172.16.10.200 port 16292
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3d19b5cf;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 107 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9053 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
SIP TIMER: #484: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3d19b5cf;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 107 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9053 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3d19b5cf;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 107 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807616 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:46:38 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK5da8b01e;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 107 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3d19b5cf;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 107 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807616 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
set_destination: set destination to 217.28.210.18, port 5060
We're at 172.16.10.200 port 16292
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK57a7ba87;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 108 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9054 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
SIP TIMER: #486: Retransmitting (no NAT) to 217.28.210.18:5060:
INVITE sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK57a7ba87;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 108 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9054 IN IP4 172.16.10.240
s=session
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK57a7ba87;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 108 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807617 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3d122a8b;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 108 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK57a7ba87;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 108 INVITE
Contact: <sip:4957772333@217.28.210.18>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 269

v=0
o=AudiocodesGW 1165807932 1165807617 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6341 IN IP4 217.28.210.18

--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 217.28.210.18:6340
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4957772333@217.28.210.18> for address/port to send to
set_destination: set destination to 217.28.210.18, port 5060
Transmitting (no NAT) to 217.28.210.18:5060:
ACK sip:4957772333@217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK620c2cc1;rport
From: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
To: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
Contact: <sip:4956273690@172.16.10.200>
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 108 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
BYE sip:4956273690@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1851306464
Max-Forwards: 70
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 3 BYE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
Reason: Q.850 ;cause=16
Content-Length: 0


--- (12 headers 0 lines) ---
Sending to 217.28.210.18 : 5060 (non-NAT)
Transmitting (no NAT) to 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.28.210.18;branch=z9hG4bKac1851306464;received=217.28.210.18
From: <sip:4957772333;cpc=unknown@TELENET>;tag=1c1165839263
To: <sip:4956273690@172.16.10.200;user=phone>;tag=as142e7761
Call-ID: 11658383702142009111129@217.28.210.18
CSeq: 3 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4956273690@172.16.10.200>
Content-Length: 0
X-CallWeaver-HangupCause: Normal Clearing


---
== Spawn extension (internal, 4956273690, 1) exited non-zero on 'SIP/4957772333-0940c8f8'
Destroying call '11658383702142009111129@217.28.210.18'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.28.210.18:5060:
OPTIONS sip:217.28.210.18 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK32ff554c;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as6ee18ded
To: <sip:217.28.210.18>
Contact: <sip:callweaver@172.16.10.200>
Call-ID: 11fa0e554a47660e17e9a522329de12a@172.16.10.200
CSeq: 102 OPTIONS
User-Agent: CallWeaver
Max-Forwards: 70
Date: Tue, 21 Apr 2009 06:47:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 217.28.210.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK32ff554c;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as6ee18ded
To: <sip:217.28.210.18>;tag=1c428156343
Call-ID: 11fa0e554a47660e17e9a522329de12a@172.16.10.200
CSeq: 102 OPTIONS
Contact: <sip:217.28.210.18>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.60A.014.009
X-Resources: telchs=24/6;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 569

v=0
o=AudiocodesGW 428166141 428165818 IN IP4 217.28.210.18
s=Phone-Call
c=IN IP4 217.28.210.18
t=0 0
m=audio 6000 RTP/AVP 4 0 8 18 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=rtcp:6001 IN IP4 217.28.210.18
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 24 lines) ---
Destroying call '11fa0e554a47660e17e9a522329de12a@172.16.10.200'
centos1*CLI> quit

2009-04-21 11:39

uni_advisor
Сообщений: 54

Re: Callweaver и факсы

Дебаги для шлюза с факсом.
centos1*CLI>sip debug sip debug peer 4956273690
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
OPTIONS sip:172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7013ec9f;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as6b2774a5
To: <sip:172.16.10.240>
Contact: <sip:callweaver@172.16.10.200>
Call-ID: 6255c8e310af7a5e1aec5d1656135047@172.16.10.200
CSeq: 102 OPTIONS
User-Agent: CallWeaver
Max-Forwards: 70
Date: Tue, 21 Apr 2009 06:52:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7013ec9f;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as6b2774a5
To: <sip:172.16.10.240>;tag=1c1629620644
Call-ID: 6255c8e310af7a5e1aec5d1656135047@172.16.10.200
CSeq: 102 OPTIONS
Contact: <sip:4956273690@172.16.10.240>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
X-Resources: telchs=1/0;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 570

v=0
o=AudiocodesGW 1629624125 1629623997 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 24 lines) ---
Destroying call '6255c8e310af7a5e1aec5d1656135047@172.16.10.200'
Apr 21 10:52:29 WARNING[3063942032]: channel.c:683 cw_channel_perform_set_t38_status: cw_channel_set_t38_status called with NULL channel at chan_sip.c:4934
-- Executing [4956273690@internal:1] Dial("SIP/4957772333-0940c8f8", "SIP/4956273690")
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7a0f8fba;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Date: Tue, 21 Apr 2009 06:52:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 9046 9046 IN IP4 172.16.10.200
s=session
c=IN IP4 172.16.10.200
t=0 0
m=audio 19884 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 4956273690
SIP TIMER: #515: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7a0f8fba;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Date: Tue, 21 Apr 2009 06:52:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 9046 9046 IN IP4 172.16.10.200
s=session
c=IN IP4 172.16.10.200
t=0 0
m=audio 19884 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7a0f8fba;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Length: 0


--- (10 headers 0 lines) ---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7a0f8fba;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Length: 0


--- (11 headers 0 lines) ---
-- SIP/4956273690-aa02 is ringing
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7a0f8fba;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Length: 0


--- (11 headers 0 lines) ---
-- SIP/4956273690-aa02 is ringing
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7a0f8fba;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456476 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:4956273690@172.16.10.240>
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7b6a9096;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 102 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
-- SIP/4956273690-aa02 answered SIP/4957772333-0940c8f8
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 14 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7fe595fe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 9046 9047 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
SIP TIMER: #518: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7fe595fe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 103 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 9046 9047 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7fe595fe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 103 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456477 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK321cb55a;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 103 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK6d825f19;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 104 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9048 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7fe595fe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 103 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456477 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
SIP TIMER: #520: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK6d825f19;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 104 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9048 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK6d825f19;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 104 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456478 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK142c93f2;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 104 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK6d825f19;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 104 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456478 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK203efe4c;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 104 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
INVITE sip:4957772333;cpc=unknown@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1693938684
Max-Forwards: 70
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 1 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 307

v=0
o=AudiocodesGW 1672456596 1672456479 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=ptime:20

--- (13 headers 13 lines) ---
Using INVITE request as basis request - 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
Sending to 172.16.10.240 : 5060 (non-NAT)
Got T.38 offer in SDP
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:52:34 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4957772333-0940c8f8 - Call ID 5730932882142009112144@217.28.210.18!
Apr 21 10:52:34 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
Apr 21 10:52:34 ERROR[3063942032]: chan_sip.c:14614 sipsock_read: We could NOT get the channel lock for SIP/4957772333-0940c8f8 - Call ID 5730932882142009112144@217.28.210.18!
Apr 21 10:52:34 ERROR[3063942032]: chan_sip.c:14615 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0
T.38 UDPTL is at port 172.16.10.200:19884...
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1693938684;received=172.16.10.240
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9049 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=image 6322 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #522: Retransmitting (no NAT) to 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1693938684;received=172.16.10.240
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9049 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=image 6322 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
ACK sip:4957772333;cpc=unknown@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1694308524
Max-Forwards: 70
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 1 ACK
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Length: 0


--- (12 headers 0 lines) ---
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3edee9d4;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9050 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Apr 21 10:52:34 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
SIP TIMER: #524: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3edee9d4;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9050 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
ACK sip:4957772333;cpc=unknown@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1694308524
Max-Forwards: 70
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 1 ACK
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Length: 0


--- (12 headers 0 lines) ---
Apr 21 10:52:34 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
Apr 21 10:52:34 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
SIP TIMER: #524: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3edee9d4;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9050 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3edee9d4;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456480 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4881abbc;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK61fec1fb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9051 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3edee9d4;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456480 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
SIP TIMER: #526: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK61fec1fb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9051 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK3edee9d4;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 105 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456480 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
SIP TIMER: #526: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK61fec1fb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9051 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK61fec1fb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456481 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK2a535000;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK61fec1fb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456481 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK4732dd9d;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK61fec1fb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456481 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK7abf7d38;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 106 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
T.38 UDPTL is at port 172.16.10.200:19884...
13 headers, 15 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9052 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=image 6322 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #528: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9052 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=image 6322 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #528: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9052 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=image 6322 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
SIP TIMER: #528: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 343

v=0
o=root 9046 9052 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=image 6322 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:122
a=T38FaxMaxDatagram:122
a=T38FaxUdpEC:t38UDPFEC

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 295

v=0
o=AudiocodesGW 1672456596 1672456482 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (12 headers 12 lines) ---
Got T.38 offer in SDP
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK507ccc8f;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1d8690e7;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 108 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9053 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 295

v=0
o=AudiocodesGW 1672456596 1672456482 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (12 headers 12 lines) ---

<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 295

v=0
o=AudiocodesGW 1672456596 1672456482 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (12 headers 12 lines) ---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK44356611;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 107 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 295

v=0
o=AudiocodesGW 1672456596 1672456482 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (12 headers 12 lines) ---
SIP TIMER: #530: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1d8690e7;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 108 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9053 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6322 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1d8690e7;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 108 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456483 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 21 10:52:35 WARNING[3063942032]: chan_sip.c:12370 handle_response_invite: RTP re-invite after T38 session not handled yet !
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK40c052fa;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 108 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
We're at 172.16.10.200 port 19884
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK371dbbfe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 109 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9054 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1d8690e7;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 108 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456483 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
SIP TIMER: #533: Retransmitting (no NAT) to 172.16.10.240:5060:
INVITE sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK371dbbfe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 109 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 9046 9054 IN IP4 217.28.210.18
s=session
c=IN IP4 217.28.210.18
t=0 0
m=audio 6320 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK371dbbfe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 109 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456484 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK6d2e0a19;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 109 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK371dbbfe;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 109 INVITE
Contact: <sip:4956273690@172.16.10.240>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Content-Type: application/sdp
Content-Length: 269

v=0
o=AudiocodesGW 1672456596 1672456484 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.16.10.240:6000
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:4956273690@172.16.10.240> for address/port to send to
set_destination: set destination to 172.16.10.240, port 5060
Transmitting (no NAT) to 172.16.10.240:5060:
ACK sip:4956273690@172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK36ccd3bb;rport
From: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
To: <sip:4956273690@172.16.10.240>;tag=1c1672435755
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 109 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
BYE sip:4957772333;cpc=unknown@172.16.10.200 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1744623512
Max-Forwards: 70
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 2 BYE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
Reason: Q.850 ;cause=16
Content-Length: 0


--- (12 headers 0 lines) ---
Sending to 172.16.10.240 : 5060 (non-NAT)
Transmitting (no NAT) to 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.240;branch=z9hG4bKac1744623512;received=172.16.10.240
From: <sip:4956273690@172.16.10.240>;tag=1c1672435755
To: "4957772333;cpc=unknown" <sip:4957772333;cpc=unknown@172.16.10.200>;tag=as577258d9
Call-ID: 409aa1445d33cacc2c39390b0f602ca1@172.16.10.200
CSeq: 2 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4957772333;cpc=unknown@172.16.10.200>
Content-Length: 0
X-CallWeaver-HangupCause: Normal Clearing


---
== Spawn extension (internal, 4956273690, 1) exited non-zero on 'SIP/4957772333-0940c8f8'
Destroying call '409aa1445d33cacc2c39390b0f602ca1@172.16.10.200'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 172.16.10.240:5060:
OPTIONS sip:172.16.10.240 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1d64adbe;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as484326c6
To: <sip:172.16.10.240>
Contact: <sip:callweaver@172.16.10.200>
Call-ID: 326ab3553bbb522c3d20715941af76c1@172.16.10.200
CSeq: 102 OPTIONS
User-Agent: CallWeaver
Max-Forwards: 70
Date: Tue, 21 Apr 2009 06:53:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
centos1*CLI>
<-- SIP read from 172.16.10.240:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK1d64adbe;rport
From: "callweaver" <sip:callweaver@172.16.10.200>;tag=as484326c6
To: <sip:172.16.10.240>;tag=1c1883686738
Call-ID: 326ab3553bbb522c3d20715941af76c1@172.16.10.200
CSeq: 102 OPTIONS
Contact: <sip:4956273690@172.16.10.240>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-112 FXS/v.5.20A.031.007
X-Resources: telchs=1/0;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 570

v=0
o=AudiocodesGW 1883690187 1883690059 IN IP4 172.16.10.240
s=Phone-Call
c=IN IP4 172.16.10.240
t=0 0
m=audio 6000 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 172.16.10.240
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

--- (14 headers 24 lines) ---
Destroying call '326ab3553bbb522c3d20715941af76c1@172.16.10.200'
2009-04-21 11:39

uni_advisor
Сообщений: 54

Re: Callweaver и факсы

В конфигурации шлюзов у обоих поставил кодеки ulaw,alaw
disallow=all
allow=ulaw
allow=alaw

2009-04-21 11:42

noize
Avatara of noize
Откуда: Москва
Сообщений: 398

Re: Callweaver и факсы

такие логи только аттачем. Думаете, кому то тут охота листать эти километры информации?
2009-04-21 11:48

uni_advisor
Сообщений: 54

Re: Callweaver и факсы

Знающие люди подскажите пожалуйста может изначально схема неправильная ? И все что нужно - купить телефонную плату Digium и туда подключить E1, исключив Медиант из цепочки. Будет ли тогда схема нормально функционировать при немодифицированном астере ?
2009-04-21 11:49

dimas
Сообщений: 866

Re: Callweaver и факсы

Просто любопытно - а не проще астериск 1.6 попробовать? У меня точно факсы по T38 в нем ходили.
2009-04-21 14:07

uni_advisor
Сообщений: 54

Re: Callweaver и факсы

Я пробовал. 1.6 при текущей схеме не работает. Зато когда поставил 1.4 астер наложил патчи asterisk-1.4.21.1-app_fax-support-t38.patch. И все факсы пошли. Конфигурации sip.conf и extensions.conf при тестах были аналогичные.
2009-04-21 15:41

dimas
Сообщений: 866

Re: Callweaver и факсы

Что значит "1.6 при текущей схеме не работает" ? не работают факсы или сам Астериск?

в 1.6 факс сделан "правильнее" чем в 1.4. Собственно факс в 1.4 это бэкпорт, то есть попытка перетащить функциональность которую добавили в 1.6 - обратно в 1.4. При этом сия задача не очень простая изначально, т.к. кроме ттого что там приложение app_fax добавлено было, там еще SIP модуль ковырялся и много чего вокруг. Cache сделал бэкпорт в какой-то момент времени, но вы догадываетесь что Digium этот бэкпорт не поддерживает... Соответственно все баги которые потом находились как в самом факсе так и в sip/T38 исправлялись только в 1.6 - в 1.4 их никто не затаскивал.

в общем если факс работал в 1.4 то в 1.6 он должен работать и подавно. Если нет - логи в студию.
2009-04-21 16:05

Vlad1983
Сообщений: 6

Re: Callweaver и факсы

По голосу факсы гоняю и всё в норме.
Даже не собираюсь с T.38 заморачиваться.
Ходят даже не шлюзах D-link правда не на всех, а на тех кто мало умничает и не пытается детектировать сигналы факса.
С Audiocodes вообще проблем никаких.
2009-04-24 10:10

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