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t38 gateway

0

Необходимо, чтобы заработал t38 gateway в Asterisk. Использую Asterisk 1.8.10 вот с этим патчем https://issues.asterisk.org/view.php?id=13405.

Схема включения:

FAX1 <> FXS <> ASTERISK1 <> IP <> ASTERISK2 <> FXS <> FAX2

Между двумя астерисками sip trunk. Настройки факса в sip.conf

t38pt_udptl = yes,redundancy(пробовал все возможные).

Экстеншн на входящие:

[fax]
exten => 118,1,NoOp()
exten => 118,n,Set(FAXOPT(gateway)=yes)
exten => 118,n,Dial(SLIC//dev/si3226x_fxs1.0,600)

Экстеншн на исходящие:

[fxs1]
exten => _7.,1,NoOp()
exten => _7.,n,Set(FAXOPT(gateway)=yes)
exten => _7.,n,Dial(SIP/trunk_2/${EXTEN:1}, 30)

Астериски с двух сторон настраиваю симметрично.

Вот часть вывода asterisk -rd:

<------------->
[Jun 14 12:30:54] VERBOSE[3309] chan_sip.c: --- (13 headers 0 lines) ---
[Jun 14 12:30:54] VERBOSE[3309] chan_sip.c: Scheduling destruction of SIP dialog '2cb2500a4dbe2f3a74ef218c33233d73@192.168.10.88' in 32000 ms (Method: REGISTER)
[Jun 14 12:30:54] NOTICE[3309] chan_sip.c: Outbound Registration: Expiry for 192.168.10.88 is 120 sec (Scheduling reregistration in 105 s)
[Jun 14 12:31:17] DEBUG[3311] config.c: Parsing /etc/asterisk/si3226x.conf
[Jun 14 12:31:17] VERBOSE[3311] chan_si3226x.c:     -- event_offhook, iface: /dev/si3226x_fxs1.0
[Jun 14 12:31:17] VERBOSE[3311] chan_si3226x.c:     --   AST_STATE_DOWN: 
[Jun 14 12:31:19] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 55
[Jun 14 12:31:19] VERBOSE[3311] chan_si3226x.c:     -- event_dtmf 7
[Jun 14 12:31:19] NOTICE[3311] chan_si3226x.c: state: 2
[Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 49
[Jun 14 12:31:20] VERBOSE[3311] chan_si3226x.c:     -- event_dtmf 1
[Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: state: 3
[Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 49
[Jun 14 12:31:20] VERBOSE[3311] chan_si3226x.c:     -- event_dtmf 1
[Jun 14 12:31:20] NOTICE[3311] chan_si3226x.c: state: 3
[Jun 14 12:31:21] NOTICE[3311] chan_si3226x.c: SLIC_GET_DTMF_ASCII: 56
[Jun 14 12:31:21] VERBOSE[3311] chan_si3226x.c:     -- event_dtmf 8
[Jun 14 12:31:21] NOTICE[3311] chan_si3226x.c: state: 3
[Jun 14 12:31:23] VERBOSE[3311] chan_si3226x.c:     -- event_digit_timer
[Jun 14 12:31:23] VERBOSE[3311] chan_si3226x.c:     --   extension exists, starting PBX 7118
[Jun 14 12:31:23] VERBOSE[3311] chan_si3226x.c:     -- Sarting audio transfer on line /dev/si3226x_fxs1.0
[Jun 14 12:31:23] DEBUG[3315] pbx.c: Launching 'NoOp'
[Jun 14 12:31:23] DEBUG[3315] pbx.c: Launching 'Set'
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun 14 12:31:23] DEBUG[3315] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun 14 12:31:23] DEBUG[3315] pbx.c: Launching 'Dial'
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Allocating new SIP dialog for 68ce19ef4f2231bd42aa84640213ea44@(null) - INVITE (No RTP)
[Jun 14 12:31:23] DEBUG[3315] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x12df758'
[Jun 14 12:31:23] DEBUG[3315] res_rtp_asterisk.c: Allocated port 15234 for RTP instance '0x12df758'
[Jun 14 12:31:23] DEBUG[3315] rtp_engine.c: RTP instance '0x12df758' is setup and ready to go
[Jun 14 12:31:23] DEBUG[3315] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x12df758'
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Setting NAT on RTP to Off
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Jun 14 12:31:23] DEBUG[3315] acl.c: Not an IPv4 nor IPv6 address, cannot get port.
[Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALEDTIME.
[Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable ANSWEREDTIME.
[Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALEDPEERNAME.
[Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALEDPEERNUMBER.
[Jun 14 12:31:23] DEBUG[3315] channel.c: Not copying variable DIALSTATUS.
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Outgoing Call for 118
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: False Text flag: False
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) 
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Audio is at 15234
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding codec 0x800 (g726) to SDP
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jun 14 12:31:23] DEBUG[3315] chan_sip.c: Initializing initreq for method INVITE - callid 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:23] VERBOSE[3315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.88:5060:
INVITE sip:118@192.168.10.88 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623
Max-Forwards: 70
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>
Contact: <sip:118@192.168.10.89:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.0
Date: Thu, 14 Jun 2012 12:31:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 862485137 862485137 IN IP4 192.168.10.89
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.89
t=0 0
m=audio 15234 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623;received=192.168.10.89
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.88:5060>
Content-Length: 0

<------------->
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: --- (11 headers 0 lines) ---
[Jun 14 12:31:24] DEBUG[3309] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' Request 102: Found
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623;received=192.168.10.89
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.88:5060>
Content-Type: application/sdp
Content-Length: 344

v=0
o=root 1997296106 1997296106 IN IP4 192.168.10.88
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.88
t=0 0
m=audio 19418 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: --- (12 headers 15 lines) ---
[Jun 14 12:31:24] DEBUG[3309] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' Request 102: Found
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: list_route: hop: <sip:118@192.168.10.88:5060>
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 0
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 0 based on m type on 0x4090f0e8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 8
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 8 based on m type on 0x4090f0e8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 3
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 3 based on m type on 0x4090f0e8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 111
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 111 based on m type on 0x4090f0e8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found RTP audio format 101
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Setting payload 101 based on m type on 0x4090f0e8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format PCMU for ID 0
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format PCMA for ID 8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format GSM for ID 3
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format G726-32 for ID 111
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 0 on 0x4090f0e8
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 3 on 0x4090f0e8
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 8 on 0x4090f0e8
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 101 on 0x4090f0e8
[Jun 14 12:31:24] DEBUG[3309] rtp_engine.c: Incorporating payload 111 on 0x4090f0e8
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x80e (gsm|ulaw|alaw|g726)
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 14 12:31:24] DEBUG[3309] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x12df758'
[Jun 14 12:31:24] VERBOSE[3309] chan_sip.c: Peer audio RTP is at port 192.168.10.88:19418
[Jun 14 12:31:24] DEBUG[3309] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x12df758'
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK36358623;received=192.168.10.89
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.88:5060>
Content-Type: application/sdp
Content-Length: 344

v=0
o=root 1997296106 1997296107 IN IP4 192.168.10.88
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.88
t=0 0
m=audio 19418 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: --- (12 headers 15 lines) ---
[Jun 14 12:31:26] DEBUG[3309] chan_sip.c: Acked pending invite 102
[Jun 14 12:31:26] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Request 102: Match Found
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 0
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 0 based on m type on 0x4090f0e8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 8
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 8 based on m type on 0x4090f0e8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 3
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 3 based on m type on 0x4090f0e8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 111
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 111 based on m type on 0x4090f0e8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found RTP audio format 101
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Setting payload 101 based on m type on 0x4090f0e8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format PCMU for ID 0
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format PCMA for ID 8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format GSM for ID 3
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format G726-32 for ID 111
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 0 on 0x4090f0e8
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 3 on 0x4090f0e8
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 8 on 0x4090f0e8
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 101 on 0x4090f0e8
[Jun 14 12:31:26] DEBUG[3309] rtp_engine.c: Incorporating payload 111 on 0x4090f0e8
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x80e (gsm|ulaw|alaw|g726)
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 14 12:31:26] DEBUG[3309] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x12df758'
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Peer audio RTP is at port 192.168.10.88:19418
[Jun 14 12:31:26] DEBUG[3309] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x12df758'
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: list_route: hop: <sip:118@192.168.10.88:5060>
[Jun 14 12:31:26] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Transmitting (no NAT) to 192.168.10.88:5060:
ACK sip:118@192.168.10.88:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK55c36014
Max-Forwards: 70
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Contact: <sip:118@192.168.10.89:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.0
Content-Length: 0


---
[Jun 14 12:31:26] DEBUG[3315] features.c: Removing dialed interfaces datastore on SIP/trunk_2-00000000 since we're bridging
[Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel /dev/si3226x_fxs1.0 to read format slin
[Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel /dev/si3226x_fxs1.0 to write format slin
[Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel SIP/trunk_2-00000000 to read format slin
[Jun 14 12:31:26] DEBUG[3315] channel.c: Set channel SIP/trunk_2-00000000 to write format slin
[Jun 14 12:31:26] VERBOSE[3315] chan_si3226x.c:     -- start audio transfer, set silence state on line /dev/si3226x_fxs1.0
[Jun 14 12:31:26] DEBUG[3315] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Jun 14 12:31:26] DEBUG[3315] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[Jun 14 12:31:26] VERBOSE[3309] chan_sip.c: Really destroying SIP dialog '2cb2500a4dbe2f3a74ef218c33233d73@192.168.10.88' Method: REGISTER
[Jun 14 12:31:31] DEBUG[3315] res_rtp_asterisk.c: Got RTCP report of 64 bytes
[Jun 14 12:31:36] DEBUG[3315] res_rtp_asterisk.c: Got RTCP report of 64 bytes
[Jun 14 12:31:38] DEBUG[3315] dsp.c: 2100 Hz done detected
[Jun 14 12:31:38] NOTICE[3315] channel.c: Dropping incompatible voice frame on SIP/trunk_2-00000000 of format slin since our native format has changed to 0x4 (ulaw)
[Jun 14 12:31:38] DEBUG[3315] chan_sip.c: Setting NAT on UDPTL to Off
[Jun 14 12:31:38] DEBUG[3315] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:38] VERBOSE[3315] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to
[Jun 14 12:31:38] VERBOSE[3315] chan_sip.c: set_destination: set destination to 192.168.10.88:5060
[Jun 14 12:31:38] DEBUG[3315] chan_sip.c: T.38 UDPTL is at 192.168.10.89 port 4551
[Jun 14 12:31:38] DEBUG[3315] chan_sip.c: Initializing already initialized SIP dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 (presumably reinvite)
[Jun 14 12:31:38] VERBOSE[3315] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.88:5060:
INVITE sip:118@192.168.10.88:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK17a5c3ac
Max-Forwards: 70
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Contact: <sip:118@192.168.10.89:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 862485137 862485138 IN IP4 192.168.10.89
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.89
t=0 0
m=image 4551 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC

---
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
INVITE sip:118@192.168.10.89:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK2ad630a1
Max-Forwards: 70
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Contact: <sip:118@192.168.10.88:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1997296106 1997296108 IN IP4 192.168.10.88
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.88
t=0 0
m=image 4541 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: --- (14 headers 11 lines) ---
[Jun 14 12:31:38] WARNING[3309] chan_sip.c: Failed to read an alternate host or port in SDP. Expect audio problems
[Jun 14 12:31:38] WARNING[3309] chan_sip.c: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 192.168.10.88:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK2ad630a1;received=192.168.10.88
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------>
[Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Got INVITE on call where we already have pending INVITE, deferring that - 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK17a5c3ac;received=192.168.10.89
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: --- (12 headers 0 lines) ---
[Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Acked pending invite 103
[Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Request 103: Match Found
[Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: Transmitting (no NAT) to 192.168.10.88:5060:
ACK sip:118@192.168.10.88:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK17a5c3ac
Max-Forwards: 70
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Contact: <sip:118@192.168.10.89:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.0
Content-Length: 0


---
[Jun 14 12:31:38] WARNING[3309] chan_sip.c: just did sched_add waitid(13) for sip_reinvite_retry for dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 in handle_response_invite
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
ACK sip:118@192.168.10.89:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK2ad630a1
Max-Forwards: 70
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Contact: <sip:118@192.168.10.88:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.0
Content-Length: 0

<------------->
[Jun 14 12:31:38] VERBOSE[3309] chan_sip.c: --- (10 headers 0 lines) ---
[Jun 14 12:31:38] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Response 102: Match Found
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
INVITE sip:118@192.168.10.89:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK21f9b33c
Max-Forwards: 70
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Contact: <sip:118@192.168.10.88:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1997296106 1997296109 IN IP4 192.168.10.88
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.88
t=0 0
m=image 4541 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: --- (14 headers 11 lines) ---
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Sending to 192.168.10.88:5060 (no NAT)
[Jun 14 12:31:40] DEBUG[3309] chan_sip.c: Initializing initreq for method INVITE - callid 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Got T.38 offer in SDP in dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Jun 14 12:31:40] DEBUG[3309] chan_sip.c: Peer T.38 UDPTL is at port 192.168.10.88:4541
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: 
<--- Transmitting (no NAT) to 192.168.10.88:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK21f9b33c;received=192.168.10.88
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.89:5060>
Content-Length: 0


<------------>
[Jun 14 12:31:40] DEBUG[3315] chan_sip.c: T.38 UDPTL is at 192.168.10.89 port 4551
[Jun 14 12:31:40] VERBOSE[3315] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 192.168.10.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK21f9b33c;received=192.168.10.88
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.89:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 862485137 862485139 IN IP4 192.168.10.89
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.89
t=0 0
m=image 4551 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC

<------------>
[Jun 14 12:31:40] DEBUG[3315] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
ACK sip:118@192.168.10.89:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.88:5060;branch=z9hG4bK78a876b1
Max-Forwards: 70
From: <sip:118@192.168.10.88>;tag=as0bead541
To: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
Contact: <sip:118@192.168.10.88:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.0
Content-Length: 0

<------------->
[Jun 14 12:31:40] VERBOSE[3309] chan_sip.c: --- (10 headers 0 lines) ---
[Jun 14 12:31:40] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Response 103: Match Found
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: True Text flag: True
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) 
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Audio is at 15234
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding codec 0x800 (g726) to SDP
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Initializing already initialized SIP dialog 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88 (presumably reinvite)
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.10.88:5060:
INVITE sip:118@192.168.10.88:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK656f8b61
Max-Forwards: 70
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Contact: <sip:118@192.168.10.89:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 862485137 862485140 IN IP4 192.168.10.89
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.89
t=0 0
m=audio 15234 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK656f8b61;received=192.168.10.89
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.88:5060>
Content-Length: 0

<------------->
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: --- (11 headers 0 lines) ---
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' Request 104: Found
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: 
<--- SIP read from UDP:192.168.10.88:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK656f8b61;received=192.168.10.89
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:118@192.168.10.88:5060>
Content-Type: application/sdp
Content-Length: 344

v=0
o=root 1997296106 1997296110 IN IP4 192.168.10.88
s=Asterisk PBX 1.8.10.0
c=IN IP4 192.168.10.88
t=0 0
m=audio 19418 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: --- (12 headers 15 lines) ---
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Acked pending invite 104
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Stopping retransmission on '5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88' of Request 104: Match Found
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 0
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 0 based on m type on 0x4090f0e8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 8
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 8 based on m type on 0x4090f0e8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 3
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 3 based on m type on 0x4090f0e8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 111
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 111 based on m type on 0x4090f0e8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found RTP audio format 101
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Setting payload 101 based on m type on 0x4090f0e8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format PCMU for ID 0
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format PCMA for ID 8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format GSM for ID 3
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format G726-32 for ID 111
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 0 on 0x4090f0e8
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 3 on 0x4090f0e8
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 8 on 0x4090f0e8
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 101 on 0x4090f0e8
[Jun 14 12:31:41] DEBUG[3309] rtp_engine.c: Incorporating payload 111 on 0x4090f0e8
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x80e (gsm|ulaw|alaw|g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x80e (gsm|ulaw|alaw|g726)
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Peer audio RTP is at port 192.168.10.88:19418
[Jun 14 12:31:41] DEBUG[3309] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x12df758'
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Jun 14 12:31:41] DEBUG[3309] chan_sip.c: Strict routing enforced for session 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: Parsing <sip:118@192.168.10.88:5060> for address/port to send to
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: set_destination: set destination to 192.168.10.88:5060
[Jun 14 12:31:41] VERBOSE[3309] chan_sip.c: Transmitting (no NAT) to 192.168.10.88:5060:
ACK sip:118@192.168.10.88:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.89:5060;branch=z9hG4bK286706cd
Max-Forwards: 70
From: "Anonymous" <sip:118@anonymous.invalid>;tag=as769d395b
To: <sip:118@192.168.10.88>;tag=as0bead541
Contact: <sip:118@192.168.10.89:5060>
Call-ID: 5ee1cde2504b0baf16e9275d71bfa57c@192.168.10.88
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.10.0
Content-Length: 0


---
[Jun 14 12:31:41] DEBUG[3315] channel.c: Got a FRAME_CONTROL (31) frame on channel SIP/trunk_2-00000000
[Jun 14 12:31:41] DEBUG[3315] channel.c: Bridge stops bridging channels /dev/si3226x_fxs1.0 and SIP/trunk_2-00000000
[Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode
[Jun 14 12:31:41] DEBUG[3315] channel.c: Generator got voice, switching to phase locked mode
[Jun 14 12:31:41] DEBUG[3315] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Jun 14 12:31:41] DEBUG[3315] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x12df758'
[Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode
[Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode
[Jun 14 12:31:41] DEBUG[3315] res_fax.c: Incompatible T.38 State for gateway mode

Дальше долго идёт T.38 State for gateway mode. Есть ещё вывод wiresharkа, но он в текстовом виде как то плохо читаемый. Есть у кого какие мысли, почему факсы не проходят?

удалить закрыть спам изменить тег редактировать

спросил 2012-06-16 19:47:58 +0400

savva Gravatar savva
19 3 2 7

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можете тут еще глянуть

http://forum.asterisk.ru/viewtopic.php?f=3&t=1774

switch ( 2012-06-17 06:47:30 +0400 )редактировать

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0

t.38 даже без gateway требует знания уровня експерт. чтото вам посоветовать сложно кроме "читайте rfc t.38 и ищите особенности своих шлюзов"

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ответил 2012-06-17 00:38:46 +0400

meral Gravatar meral flag of Ukraine
21228 23 18 169
http://pro-sip.net/

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Задан: 2012-06-16 19:47:58 +0400

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