-- Executing [379641@sip-isdn-in:2] Dial("SIP/10000-00000008", "SIP/338") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 192.168.111.5 port 16764 Adding codec 0x8 (alaw) to SDP Adding codec 0x1000 (g722) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x8000 (slin16) to SDP Reliably Transmitting (no NAT) to 192.168.110.171:5060: INVITE sip:338@192.168.110.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK499f4d0f;rport Max-Forwards: 70 From: "0892442191771" ;tag=as035c3f3c To: Contact: Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.6-rc2 Date: Fri, 12 Mar 2010 11:23:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 515 v=0 o=root 1070143141 1070143141 IN IP4 192.168.111.5 s=Asterisk PBX 1.6.2.6-rc2 c=IN IP4 192.168.111.5 t=0 0 m=audio 16764 RTP/AVP 8 9 3 0 112 5 10 7 18 97 111 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 338 asterisk*CLI> <--- SIP read from UDP:192.168.110.171:5060 ---> SIP/2.0 100 Trying To: From: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK499f4d0f Server: Linksys/SPA2102-5.2.10 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from UDP:192.168.110.171:5060 ---> SIP/2.0 180 Ringing To: ;tag=3566d0f0ccbebd2ci0 From: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK499f4d0f Contact: 338 Server: Linksys/SPA2102-5.2.10 Remote-Party-ID: 338 ;screen=yes;party=called Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/338-00000009 is ringing <--- SIP read from UDP:192.168.110.171:5060 ---> SIP/2.0 200 OK To: ;tag=3566d0f0ccbebd2ci0 From: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK499f4d0f Contact: 338 Server: Linksys/SPA2102-5.2.10 Remote-Party-ID: 338 ;screen=yes;party=called Content-Length: 263 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 51619549 51619549 IN IP4 192.168.110.171 s=- c=IN IP4 192.168.110.171 t=0 0 m=audio 16448 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 100 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.110.171:16448 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.110.171, port 5060 Transmitting (no NAT) to 192.168.110.171:5060: ACK sip:338@192.168.110.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK68b736d0;rport Max-Forwards: 70 From: "0892442191771" ;tag=as035c3f3c To: ;tag=3566d0f0ccbebd2ci0 Contact: Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.6-rc2 Content-Length: 0 --- -- SIP/338-00000009 answered SIP/10000-00000008 asterisk*CLI> <--- SIP read from UDP:192.168.110.171:5060 ---> INVITE sip:0892442191771@192.168.111.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.171:5060;branch=z9hG4bK-998e137c From: ;tag=3566d0f0ccbebd2ci0 To: "0892442191771" ;tag=as035c3f3c Remote-Party-ID: 338 ;screen=yes;party=called Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 101 INVITE Max-Forwards: 70 Contact: 338 Expires: 30 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 275 Content-Type: application/sdp v=0 o=- 51621254 51621254 IN IP4 192.168.110.171 s=- c=IN IP4 192.168.110.171 t=0 0 m=image 16448 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (13 headers 12 lines) --- Sending to 192.168.110.171 : 5060 (no NAT) Got T.38 offer in SDP in dialog 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. asterisk*CLI> <--- Transmitting (no NAT) to 192.168.110.171:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.110.171:5060;branch=z9hG4bK-998e137c;received=192.168.110.171 From: ;tag=3566d0f0ccbebd2ci0 To: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 101 INVITE Server: Asterisk PBX 1.6.2.6-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> asterisk*CLI> <--- Reliably Transmitting (no NAT) to 192.168.110.171:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.110.171:5060;branch=z9hG4bK-998e137c;received=192.168.110.171 From: ;tag=3566d0f0ccbebd2ci0 To: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 101 INVITE Server: Asterisk PBX 1.6.2.6-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 276 v=0 o=root 1070143141 1070143142 IN IP4 192.168.111.5 s=Asterisk PBX 1.6.2.6-rc2 c=IN IP4 192.168.111.5 t=0 0 m=image 4445 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:2400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:385 a=T38FaxUdpEC:t38UDPRedundancy <------------> asterisk*CLI> <--- SIP read from UDP:192.168.110.171:5060 ---> ACK sip:0892442191771@192.168.111.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.171:5060;branch=z9hG4bK-1dc782c7 From: ;tag=3566d0f0ccbebd2ci0 To: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 101 ACK Max-Forwards: 70 Contact: 338 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 0 Reliably Transmitting (no NAT) to 192.168.110.171:5060: OPTIONS sip:338@192.168.110.171:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK1661a8ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as3be02f37 To: Contact: Call-ID: 5d6ee339698dd1cc1f1926e56cc22975@192.168.111.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.6-rc2 Date: Fri, 12 Mar 2010 11:24:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP:192.168.110.171:5060 ---> SIP/2.0 486 Busy Here To: ;tag=d2be8260b3619590i0 From: "asterisk" ;tag=as3be02f37 Call-ID: 5d6ee339698dd1cc1f1926e56cc22975@192.168.111.5 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.111.5:5060;branch=z9hG4bK1661a8ed Server: Linksys/SPA2102-5.2.10 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '5d6ee339698dd1cc1f1926e56cc22975@192.168.111.5' Method: OPTIONS asterisk*CLI> <--- SIP read from UDP:192.168.110.171:5060 ---> BYE sip:0892442191771@192.168.111.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.171:5060;branch=z9hG4bK-390a7448 From: ;tag=3566d0f0ccbebd2ci0 To: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 BYE Max-Forwards: 70 User-Agent: Linksys/SPA2102-5.2.10 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.110.171 : 5060 (no NAT) asterisk*CLI> <--- Transmitting (no NAT) to 192.168.110.171:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.110.171:5060;branch=z9hG4bK-390a7448;received=192.168.110.171 From: ;tag=3566d0f0ccbebd2ci0 To: "0892442191771" ;tag=as035c3f3c Call-ID: 11bc14a26905a0c86769109c3a6bc181@192.168.111.5 CSeq: 102 BYE Server: Asterisk PBX 1.6.2.6-rc2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (sip-isdn-in, 379641, 2) exited non-zero on 'SIP/10000-00000008'