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В чем проблема?

Пытаюсь пробросить звонок из транка в другой транк
impag
Сообщений: 8

В чем проблема?

Я начинаю осваивать Asterisk, поэтому прошу помочь в чтении вебага и подсказать в чем проблема.
Получаю звонок с одного SIP-транка, и после донабора номера, хочу прокинуть его в другой SIP-транк.

Что срабатывает:
С SIP-транка дозвон до моего * происходт. Выдается приветствие и дозвонившийся вводит доп.номер (444) после чего идет дозвон по второму транку на номер 74957754477. Слышится ответ (примерно 0.5 сек.) и дается отбой. Из-за чего? в чем проблема и где мне еще что-то донастроить?
При этом с внутреннего номера звонок на номер 74957754477 уходит нормально.

extensions.conf
[from-trunk]
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,Wait,1
exten => s,n,NoOp(Call from: ${CALLERID})
exten => s,n,Answer()
exten => s,n,Wait,1
exten => s,n,Background(enter-ext-of-person)
exten => s,n,WaitExten()
exten => 444,1,Dial(SIP/SIPNet/74957754477)
exten => i,1,Playback(pbx-invalid)
exten => i,n,Hangup()
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()


debag
<--- SIP read from 92.46.61.21:5060 --->
INVITE sip:491111@92.46.153.176 SIP/2.0
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKj35l4o109om17gsfl6o0.1
From: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
To: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 292914431 INVITE
Contact: <sip:87112413635@92.46.61.21:5060;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 202

v=0
o=BroadWorks 4836440 1 IN IP4 92.46.61.21
s=-
c=IN IP4 92.46.61.21
t=0 0
m=audio 14656 RTP/AVP 18 8 13 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no

<------------->
--- (13 headers 10 lines) ---
Sending to 92.46.61.21 : 5060 (no NAT)
Using INVITE request as basis request - BW103813756140409-1813513925@10.14.0.2
Found peer 'krg.telecom'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 13
Found RTP audio format 101
Peer audio RTP is at port 92.46.61.21:14656
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 92.46.61.21:14656
Looking for 491111 in from-trunk-krg (domain 92.46.153.176)
list_route: hop: <sip:87112413635@92.46.61.21:5060;transport=udp>

<--- Transmitting (no NAT) to 92.46.61.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKj35l4o109om17gsfl6o0.1;received=92.46.61.21
From: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
To: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 292914431 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:491111@92.46.153.176>
Content-Length: 0


<------------>
-- Executing [491111@from-trunk-krg:1] NoOp("SIP/3240869744-09070df8", "Received incoming SIP connection from unknown peer to 491111") in new stack
-- Executing [491111@from-trunk-krg:2] Set("SIP/3240869744-09070df8", "DID=491111") in new stack
-- Executing [491111@from-trunk-krg:3] Goto("SIP/3240869744-09070df8", "s|1") in new stack
-- Goto (from-trunk-krg,s,1)
-- Executing [s@from-trunk-krg:1] Wait("SIP/3240869744-09070df8", "1") in new stack
-- Executing [s@from-trunk-krg:2] NoOp("SIP/3240869744-09070df8", "Call from: ") in new stack
-- Executing [s@from-trunk-krg:3] Answer("SIP/3240869744-09070df8", "") in new stack
Audio is at 92.46.153.176 port 15862
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 92.46.61.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKj35l4o109om17gsfl6o0.1;received=92.46.61.21
From: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
To: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 292914431 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:491111@92.46.153.176>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 2973 2973 IN IP4 92.46.153.176
s=session
c=IN IP4 92.46.153.176
t=0 0
m=audio 15862 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/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 - - - -
a=ptime:20
a=sendrecv

<------------>
-- Executing [s@from-trunk-krg:4] Wait("SIP/3240869744-09070df8", "1") in new stack
*CLI>
<--- SIP read from 92.46.61.21:5060 --->
ACK sip:491111@92.46.153.176 SIP/2.0
Via: SIP/2.0/UDP 92.46.61.21:5060;branch=z9hG4bKljoqqj20c8d0gg8ad101.1
From: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
To: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 292914431 ACK
Contact: <sip:87112413635@92.46.61.21:5060;transport=udp>
Max-Forwards: 69
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- Executing [s@from-trunk-krg:5] BackGround("SIP/3240869744-09070df8", "enter-ext-of-person") in new stack
-- <SIP/3240869744-09070df8> Playing 'enter-ext-of-person' (language 'en')
Reliably Transmitting (NAT) to 10.50.16.159:5060:
OPTIONS sip:2001@10.50.16.159:5060 SIP/2.0
Via: SIP/2.0/UDP 10.50.16.155:5060;branch=z9hG4bK61c6bdbd;rport
From: "Unknown" <sip:Unknown@10.50.16.155>;tag=as2cf32296
To: <sip:2001@10.50.16.159:5060>
Contact: <sip:Unknown@10.50.16.155>
Call-ID: 5505c3675a650c2e004348fc1f25d1b1@10.50.16.155
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 14 Apr 2009 04:38:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
*CLI>
<--- SIP read from 10.50.16.159:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.16.155:5060;branch=z9hG4bK61c6bdbd
To: <sip:2001@10.50.16.159:5060>
From: "Unknown" <sip:Unknown@10.50.16.155>;tag=as2cf32296
Call-ID: 5505c3675a650c2e004348fc1f25d1b1@10.50.16.155
CSeq: 102 OPTIONS
Supported: timer,100rel
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL
Contact: <sip:2001@10.50.16.159:5060>
Content-Length: 0
Accept-Encoding: identity
Accept-Language: en


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '5505c3675a650c2e004348fc1f25d1b1@10.50.16.155' Method: OPTIONS
Reliably Transmitting (NAT) to 10.50.7.158:57822:
OPTIONS sip:2000@10.50.7.158:57822;rinstance=a73f88643ec173a9 SIP/2.0
Via: SIP/2.0/UDP 10.50.16.155:5060;branch=z9hG4bK6ef7c5ad;rport
From: "Unknown" <sip:Unknown@10.50.16.155>;tag=as7cea0744
To: <sip:2000@10.50.7.158:57822;rinstance=a73f88643ec173a9>
Contact: <sip:Unknown@10.50.16.155>
Call-ID: 4d3165e27e8dde5d438188260f9129f8@10.50.16.155
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 14 Apr 2009 04:38:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
*CLI>
<--- SIP read from 10.50.7.158:57822 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.16.155:5060;branch=z9hG4bK6ef7c5ad;rport=5060
Contact: <sip:10.50.7.158:57822>
To: <sip:2000@10.50.7.158:57822;rinstance=a73f88643ec173a9>;tag=5a5f895a
From: "Unknown"<sip:Unknown@10.50.16.155>;tag=as7cea0744
Call-ID: 4d3165e27e8dde5d438188260f9129f8@10.50.16.155
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '4d3165e27e8dde5d438188260f9129f8@10.50.16.155' Method: OPTIONS
*CLI>
<--- SIP read from 92.46.158.0:18801 --->



<------------->
*CLI>
<--- SIP read from 10.50.7.158:57822 --->



<------------->
-- Executing [s@from-trunk-krg:6] WaitExten("SIP/3240869744-09070df8", "") in new stack
*CLI>
*CLI>
*CLI>
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK64541ec3;rport
From: <sip:7496786@sipnet.ru>;tag=as4a55acef
To: <sip:7496786@sipnet.ru>
Call-ID: 7470912448f746bb5f26f1b06e514568@sipnet.ru
CSeq: 133 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="7496786", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="0E6C49F6AAB7190812FB", response="2d3ab5fbeacf92fbabc6d953f1b982c7", opaque="opaqueData", qop=auth, cnonce="5f49d212", nc=00000002
Expires: 120
Contact: <sip:7496786@92.46.153.176>
Event: registration
Content-Length: 0


---
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK64541ec3;rport=5060
From: <sip:7496786@sipnet.ru>;tag=as4a55acef
To: <sip:7496786@sipnet.ru>;tag=BBD68CDE
Call-ID: 7470912448f746bb5f26f1b06e514568@sipnet.ru
CSeq: 133 REGISTER
Expires: 120
Contact: <sip:7496786@92.46.153.176>;expires=120
Event: registration
Date: Tue, 14 Apr 2009 04:38:20 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/5.2.12
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Scheduling destruction of SIP dialog '7470912448f746bb5f26f1b06e514568@sipnet.ru' in 32000 ms (Method: REGISTER)

*CLI>
<--- SIP read from 92.46.158.0:18801 --->
SUBSCRIBE sip:2010@my.sip.server SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:18801;branch=z9hG4bK-d8754z-63350740555b8944-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2010@92.46.158.0:18801>
To: "2010"<sip:2010@my.sip.server>
From: "2010"<sip:2010@my.sip.server>;tag=e078e979
Call-ID: YjI0YjMzNDVmNWM4YjcwYmUzODFiNDYxNWM2NjgyOGQ.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Event: message-summary
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 92.46.158.0 : 18801 (NAT)
Found peer '2010'

<--- Transmitting (NAT) to 92.46.158.0:18801 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:18801;branch=z9hG4bK-d8754z-63350740555b8944-1---d8754z-;received=92.46.158.0;rport=18801
From: "2010"<sip:2010@my.sip.server>;tag=e078e979
To: "2010"<sip:2010@my.sip.server>;tag=as2d687662
Call-ID: YjI0YjMzNDVmNWM4YjcwYmUzODFiNDYxNWM2NjgyOGQ.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="28ac4baf"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YjI0YjMzNDVmNWM4YjcwYmUzODFiNDYxNWM2NjgyOGQ.' in 12544 ms (Method: SUBSCRIBE)
*CLI>
<--- SIP read from 92.46.158.0:18801 --->
SUBSCRIBE sip:2010@my.sip.server SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:18801;branch=z9hG4bK-d8754z-0f37b57a7f309229-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2010@92.46.158.0:18801>
To: "2010"<sip:2010@my.sip.server>
From: "2010"<sip:2010@my.sip.server>;tag=e078e979
Call-ID: YjI0YjMzNDVmNWM4YjcwYmUzODFiNDYxNWM2NjgyOGQ.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Authorization: Digest username="2010",realm="asterisk",nonce="28ac4baf",uri="sip:2010@my.sip.server",response="31dce98c18e676be084d6e685a5a68a9",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 92.46.158.0 : 18801 (NAT)
Found peer '2010'
Looking for 2010 in from-internal (domain my.sip.server)
Scheduling destruction of SIP dialog 'YjI0YjMzNDVmNWM4YjcwYmUzODFiNDYxNWM2NjgyOGQ.' in 310000 ms (Method: SUBSCRIBE)

<--- Transmitting (NAT) to 92.46.158.0:18801 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:18801;branch=z9hG4bK-d8754z-0f37b57a7f309229-1---d8754z-;received=92.46.158.0;rport=18801
From: "2010"<sip:2010@my.sip.server>;tag=e078e979
To: "2010"<sip:2010@my.sip.server>;tag=as2d687662
Call-ID: YjI0YjMzNDVmNWM4YjcwYmUzODFiNDYxNWM2NjgyOGQ.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 300
Contact: <sip:2010@92.46.153.176>;expires=300
Content-Length: 0


<------------>
*CLI>
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK3b14b16c;rport
From: <sip:1214931@sipnet.ru>;tag=as01799d8b
To: <sip:1214931@sipnet.ru>
Call-ID: 069c523e33c9cab814c330d42e1d3a0f@sipnet.ru
CSeq: 133 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="1214931", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="0E6C49F6AAB7190812FB", response="ee9ba06e74c9e8655de23cec5b3b22d0", opaque="opaqueData", qop=auth, cnonce="2f0dabe9", nc=00000002
Expires: 120
Contact: <sip:1214931@92.46.153.176>
Event: registration
Content-Length: 0


---
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK3b14b16c;rport=5060
From: <sip:1214931@sipnet.ru>;tag=as01799d8b
To: <sip:1214931@sipnet.ru>;tag=DD63B1AB
Call-ID: 069c523e33c9cab814c330d42e1d3a0f@sipnet.ru
CSeq: 133 REGISTER
Expires: 120
Contact: <sip:1214931@92.46.153.176>;expires=120
Event: registration
Date: Tue, 14 Apr 2009 04:38:21 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/5.2.12
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Scheduling destruction of SIP dialog '069c523e33c9cab814c330d42e1d3a0f@sipnet.ru' in 32000 ms (Method: REGISTER)

*CLI>
*CLI>
*CLI>
== CDR updated on SIP/3240869744-09070df8
-- Executing [444@from-trunk-krg:1] Dial("SIP/3240869744-09070df8", "SIP/SIPNet/74957754477|20|f") in new stack
Audio is at 92.46.153.176 port 18288
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:74957754477@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK5b73ac88;rport
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>
Contact: <sip:7496786@92.46.153.176>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 14 Apr 2009 04:38:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2973 2973 IN IP4 92.46.153.176
s=session
c=IN IP4 92.46.153.176
t=0 0
m=audio 18288 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called SIPNet/74957754477
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK5b73ac88;rport=5060
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 102 INVITE
Server: CommuniGatePro/5.2.12
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 401 Authentication required
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK5b73ac88;rport=5060
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=B4F5785E
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="etc.tario.ru",nonce="54C01A0F659A74E99FE0",opaque="opaqueData",qop="auth",algorithm=MD5
Server: CommuniGatePro/5.2.12
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:74957754477@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK5b73ac88;rport
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=B4F5785E
Contact: <sip:7496786@92.46.153.176>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 92.46.153.176 port 18288
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:74957754477@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK7427a9c3;rport
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>
Contact: <sip:7496786@92.46.153.176>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="7496786", realm="etc.tario.ru", algorithm=MD5, uri="sip:74957754477@sipnet.ru", nonce="54C01A0F659A74E99FE0", response="51ec61406ad35751d88d07d2bbdeaea1", opaque="opaqueData", qop=auth, cnonce="03f379a8", nc=00000001
Date: Tue, 14 Apr 2009 04:38:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2973 2974 IN IP4 92.46.153.176
s=session
c=IN IP4 92.46.153.176
t=0 0
m=audio 18288 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK7427a9c3;rport=5060
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 INVITE
Server: CommuniGatePro/5.2.12
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 92.46.153.176:5060;rport=5060;branch=z9hG4bK7427a9c3
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.72:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.11
Content-Length: 179

v=0
o=Tario-Softswitch 11391 101 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 20936 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:20936
Found audio description format PCMA for ID 8
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:20936
-- SIP/SIPNet-09052878 is making progress passing it to SIP/3240869744-09070df8
set_destination: Parsing <sip:87112413635@92.46.61.21:5060;transport=udp> for address/port to send to
set_destination: set destination to 92.46.61.21, port 5060
Audio is at 92.46.153.176 port 15862
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 92.46.61.21:5060:
INVITE sip:87112413635@92.46.61.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK40bee308;rport
From: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
To: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
Contact: <sip:491111@92.46.153.176>
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2973 2974 IN IP4 212.53.40.81
s=session
c=IN IP4 212.53.40.81
t=0 0
m=audio 20936 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
*CLI>
<--- SIP read from 92.46.61.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;received=92.46.153.176;branch=z9hG4bK40bee308;rport=5060
From: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
To: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: multipart/mixed,application/media_control+xml,application/sdp
Contact: <sip:87112413635@92.46.61.21:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 406

v=0
o=BroadWorks 4836440 2 IN IP4 92.46.61.21
s=-
c=IN IP4 92.46.61.21
t=0 0
a=sqn: 0
a=cdsc: 1 audio RTP/AVP 8 0 108 18 13 101
a=cpar: a=rtpmap:108 AAL2-G726-32/8000
a=cpar: a=rtpmap:101 telephone-event/8000
a=cpar: a=fmtp:101 0-15
a=cpar: a=ptime:10
a=cpar: a=ptime:20
a=cpar: a=fmtp:18 annexb=yes
m=audio 14656 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 92.46.61.21:14656
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 92.46.61.21:14656
set_destination: Parsing <sip:87112413635@92.46.61.21:5060;transport=udp> for address/port to send to
set_destination: set destination to 92.46.61.21, port 5060
Transmitting (no NAT) to 92.46.61.21:5060:
ACK sip:87112413635@92.46.61.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK1858bb41;rport
From: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
To: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
Contact: <sip:491111@92.46.153.176>
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 92.46.153.176:5060;rport=5060;branch=z9hG4bK7427a9c3
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.72:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.11
Content-Length: 179

v=0
o=Tario-Softswitch 11391 101 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 20936 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:20936
Found audio description format PCMA for ID 8
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:20936
-- SIP/SIPNet-09052878 is making progress passing it to SIP/3240869744-09070df8

<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 92.46.153.176:5060;rport=5060;branch=z9hG4bK7427a9c3
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.72:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.11
Content-Length: 179

v=0
o=Tario-Softswitch 11391 101 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 20936 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:20936
Found audio description format PCMA for ID 8
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:20936
-- SIP/SIPNet-09052878 is ringing
-- SIP/SIPNet-09052878 is making progress passing it to SIP/3240869744-09070df8

<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;rport=5060;branch=z9hG4bK7427a9c3
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.72:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 INVITE
Contact: <sip:proc-5523192@212.53.35.244>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS
Server: TarioSoftswitch/3.2.11
Content-Length: 179

v=0
o=Tario-Softswitch 11391 101 IN IP4 212.53.40.81
s=SIP Call
c=IN IP4 212.53.40.81
t=0 0
m=audio 20936 RTP/AVP 8
c=IN IP4 212.53.40.81
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (14 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.81:20936
Found audio description format PCMA for ID 8
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.53.40.81:20936
list_route: hop: <sip:212.53.40.40:5060;lr>
list_route: hop: <sip:212.53.40.72:5060;lr>
list_route: hop: <sip:2577320-212.53.40.72.dialog.cgatepro;lr>
list_route: hop: <sip:212.53.35.244:5060;lr>
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:proc-5523192@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK1fc9e9d7;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.72:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Contact: <sip:7496786@92.46.153.176>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/SIPNet-09052878 answered SIP/3240869744-09070df8
-- Native bridging SIP/3240869744-09070df8 and SIP/SIPNet-09052878
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Audio is at 92.46.153.176 port 18288
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:proc-5523192@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK1bc6c3ee;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.72:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Contact: <sip:7496786@92.46.153.176>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 2973 2975 IN IP4 92.46.61.21
s=session
c=IN IP4 92.46.61.21
t=0 0
m=audio 14656 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK1bc6c3ee;rport=5060
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 104 INVITE
Server: CommuniGatePro/5.2.12
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 92.46.153.176:5060;rport=5060;branch=z9hG4bK1bc6c3ee
From: <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 104 INVITE
Server: TarioSoftswitch/3.2.11
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:proc-5523192@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK1bc6c3ee;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.72:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Contact: <sip:7496786@92.46.153.176>
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '2dc46e657a2659e90187824f0cf1910c@sipnet.ru' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
BYE sip:proc-5523192@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK26e0d455;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.72:5060;lr>,<sip:2577320-212.53.40.72.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "87112413635" <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="7496786", realm="etc.tario.ru", algorithm=MD5, uri="sip:proc-5523192@212.53.35.244", nonce="54C01A0F659A74E99FE0", response="6acdcc2902ef88590626646273be6afe", opaque="opaqueData", qop=auth, cnonce="5f43beec", nc=00000002
Content-Length: 0


---
== Spawn extension (from-trunk-krg, 444, 1) exited non-zero on 'SIP/3240869744-09070df8'
-- Executing [h@from-trunk-krg:1] NoOp("SIP/3240869744-09070df8", "Received incoming SIP connection from unknown peer to h") in new stack
-- Executing [h@from-trunk-krg:2] Set("SIP/3240869744-09070df8", "DID=s") in new stack
-- Executing [h@from-trunk-krg:3] Goto("SIP/3240869744-09070df8", "s|1") in new stack
-- Goto (from-trunk-krg,s,1)
-- Executing [s@from-trunk-krg:1] Wait("SIP/3240869744-09070df8", "1") in new stack
== Spawn extension (from-trunk-krg, s, 1) exited non-zero on 'SIP/3240869744-09070df8'
Scheduling destruction of SIP dialog 'BW103813756140409-1813513925@10.14.0.2' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:87112413635@92.46.61.21:5060;transport=udp> for address/port to send to
set_destination: set destination to 92.46.61.21, port 5060
Reliably Transmitting (no NAT) to 92.46.61.21:5060:
BYE sip:87112413635@92.46.61.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK04c700ef;rport
From: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
To: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
*CLI>
<--- SIP read from 92.46.61.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;received=92.46.153.176;branch=z9hG4bK04c700ef;rport=5060
From: "John NINA"<sip:3240869744@sip.server2;ep=92.46.153.176:5060;fw=92.46.153.176:5060>;tag=as7d87c4cd
To: <sip:87112413635@10.14.0.2;user=phone>;tag=1151171970-1239683893756-
Call-ID: BW103813756140409-1813513925@10.14.0.2
CSeq: 103 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'BW103813756140409-1813513925@10.14.0.2' Method: ACK
*CLI>
<--- SIP read from 212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;rport=5060;branch=z9hG4bK26e0d455
From: <sip:7496786@sipnet.ru>;tag=as3eded186
To: <sip:74957754477@sipnet.ru>;tag=4625f91a-5523192
Call-ID: 2dc46e657a2659e90187824f0cf1910c@sipnet.ru
CSeq: 105 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS
Server: TarioSoftswitch/3.2.11
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2dc46e657a2659e90187824f0cf1910c@sipnet.ru' Method: INVITE
Reliably Transmitting (NAT) to 92.46.158.0:18801:
OPTIONS sip:2010@92.46.158.0:18801;rinstance=0c05772193e57e91 SIP/2.0
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK6ed5feb8;rport
From: "Unknown" <sip:Unknown@92.46.153.176>;tag=as685effee
To: <sip:2010@92.46.158.0:18801;rinstance=0c05772193e57e91>
Contact: <sip:Unknown@92.46.153.176>
Call-ID: 12c46de91a28a192797273a4599c68c6@92.46.153.176
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 14 Apr 2009 04:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
*CLI>
<--- SIP read from 92.46.158.0:18801 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.46.153.176:5060;branch=z9hG4bK6ed5feb8;rport=5060
Contact: <sip:192.168.1.3:18801>
To: <sip:2010@92.46.158.0:18801;rinstance=0c05772193e57e91>;tag=5e529531
From: "Unknown"<sip:Unknown@92.46.153.176>;tag=as685effee
Call-ID: 12c46de91a28a192797273a4599c68c6@92.46.153.176
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '12c46de91a28a192797273a4599c68c6@92.46.153.176' Method: OPTIONS
*CLI>
2009-04-14 09:04

Alekz
Avatara of Alekz
Откуда: Санкт-Петербург
Сообщений: 931

Re: В чем проблема?

По-пробуйте "canreinvite = no" для пира sipnet.
Создам аварийную ситуацию. Дорого. На долго =)
2009-04-14 09:51

impag
Сообщений: 8

Re: В чем проблема?

Большое спасибо, все заработало.
Вопрос закрыт.
2009-04-14 10:00

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