Teles -> Teles, from Asterisk Side [Mar 25 21:33:54] WARNING[18787]: chan_sip.c:1975 retrans_pkt: Maximum retries exceeded on transmission 258622691757381385783566402067@ip-teles-a for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. -- Executing [431311257433@Incoming:1] Set("SIP/ip-teles-a-18b01b30", "CDR(userfield)=431311257433") in new stack ... number manipulation... -- Executing [431311257433@Incoming:18] Dial("SIP/ip-teles-a-18b01b30", "SIP/33@ip-teles-b:5060") in new stack Audio is at ip-asterisk port 13850 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to ip-teles-b:5060: INVITE sip:33@ip-teles-b SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK6d4cece0;rport From: "00436642014115" ;tag=as4118c13f To: Contact: Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 102 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "00436642014115" ;privacy=off;screen=yes Date: Wed, 25 Mar 2009 20:34:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 18769 18769 IN IP4 ip-asterisk s=session c=IN IP4 ip-asterisk t=0 0 m=audio 13850 RTP/AVP 18 101 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 --- -- Called 33@ip-teles-b:5060 DataNode1*CLI> <--- SIP read from ip-teles-b:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK6d4cece0;rport From: "00436642014115" ;tag=as4118c13f To: Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER User-Agent: TELES.VoIPBOX 13.0k 895 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- DataNode1*CLI> <--- SIP read from ip-teles-b:5060 ---> SIP/2.0 183 Session progress Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK6d4cece0;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Type: application/sdp Content-Length: 220 v=0 o=- 48 1 IN IP4 ip-teles-b s=- c=IN IP4 ip-teles-b t=0 0 m=audio 29000 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port ip-teles-b:29000 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port ip-teles-b:29000 -- SIP/ip-teles-b:5060-18aff970 is making progress passing it to SIP/ip-teles-a-18b01b30 DataNode1*CLI> <--- SIP read from ip-teles-b:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK6d4cece0;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Type: application/sdp Content-Length: 220 v=0 o=- 48 1 IN IP4 ip-teles-b s=- c=IN IP4 ip-teles-b t=0 0 m=audio 29000 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port ip-teles-b:29000 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port ip-teles-b:29000 -- SIP/ip-teles-b:5060-18aff970 is ringing -- SIP/ip-teles-b:5060-18aff970 is making progress passing it to SIP/ip-teles-a-18b01b30 DataNode1*CLI> <--- SIP read from ip-teles-b:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK6d4cece0;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Type: application/sdp ontent-Length: 220 v=0 o=- 48 1 IN IP4 ip-teles-b s=- c=IN IP4 ip-teles-b t=0 0 m=audio 29000 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port ip-teles-b:29000 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port ip-teles-b:29000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-b, port 5060 Transmitting (no NAT) to ip-teles-b:5060: ACK sip:33@ip-teles-b SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK607091d8;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Contact: Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 102 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "00436642014115" ;privacy=off;screen=yes Content-Length: 0 --- -- SIP/ip-teles-b:5060-18aff970 answered SIP/ip-teles-a-18b01b30 -- Native bridging SIP/ip-teles-a-18b01b30 and SIP/ip-teles-b:5060-18aff970 set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-b, port 5060 Audio is at ip-asterisk port 13850 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to ip-teles-b:5060: INVITE sip:33@ip-teles-b SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK5721b9f2;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Contact: Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 103 INVITE User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "00436642014115" ;privacy=off;screen=yes 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: 267 v=0 o=root 18769 18770 IN IP4 ip-teles-a s=session c=IN IP4 ip-teles-a t=0 0 m=audio 29000 RTP/AVP 18 101 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 --- DataNode1*CLI> <--- SIP read from ip-teles-b:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK5721b9f2;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 103 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Content-Type: application/sdp ontent-Length: 220 v=0 o=- 48 1 IN IP4 ip-teles-b s=- c=IN IP4 ip-teles-b t=0 0 m=audio 29000 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port ip-teles-b:29000 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port ip-teles-b:29000 set_destination: Parsing for address/port to send to set_destination: set destination to ip-teles-b, port 5060 Transmitting (no NAT) to ip-teles-b:5060: ACK sip:33@ip-teles-b SIP/2.0 Via: SIP/2.0/UDP ip-asterisk:5060;branch=z9hG4bK5e05242a;rport From: "00436642014115" ;tag=as4118c13f To: ;tag=418190883705800553448616699450 Contact: Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 103 ACK User-Agent: GlobalCall+T.38 Max-Forwards: 70 Remote-Party-ID: "00436642014115" ;privacy=off;screen=yes Content-Length: 0 --- [Mar 25 21:34:31] WARNING[18787]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. DataNode1*CLI> <--- SIP read from ip-teles-b:5060 ---> BYE sip:00436642014115@ip-asterisk SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP ip-teles-b:5060;rport;branch=z9hG4bK74801384821596328261609 From: ;tag=418190883705800553448616699450 To: "00436642014115" ;tag=as4118c13f Contact: User-Agent: TELES.VoIPBOX 13.0k 895 Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 104 BYE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to ip-teles-b : 5060 (NAT) DataNode1*CLI> <--- Transmitting (NAT) to ip-teles-b:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP ip-teles-b:5060;branch=z9hG4bK74801384821596328261609;received=ip-teles-b;rport=5060 From: ;tag=418190883705800553448616699450 To: "00436642014115" ;tag=as4118c13f Call-ID: 05e9b1ed5f9778182590f43e2e798822@ip-asterisk CSeq: 104 BYE User-Agent: GlobalCall+T.38 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> == Spawn extension (Incoming, 431311257433, 18) exited non-zero on 'SIP/ip-teles-a-18b01b30' Really destroying SIP dialog '05e9b1ed5f9778182590f43e2e798822@ip-asterisk' Method: BYE DataNode1*CLI>