We're at a.b.c.d port 13928 Answering with preferred capability 0x1 (g723) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:14151234567@p.r.s.t SIP/2.0 Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK5852b69f From: "2071234567" ;tag=as4e6afb66 To: Contact: Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Mar 2005 17:34:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 v=0 o=root 23129 23129 IN IP4 a.b.c.d s=session c=IN IP4 a.b.c.d t=0 0 m=audio 13928 RTP/AVP 4 101 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to p.r.s.t:5060 Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK5852b69f From: "2071234567" ;tag=as4e6afb66 To: Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d CSeq: 102 INVITE Server: SIPStructure/4.0.5 Content-Length: 0 8 headers, 0 lines Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK5852b69f From: "2071234567" ;tag=as4e6afb66 To: ;tag=568D5E5C-1ACD Date: Fri, 04 Mar 2005 17:34:08 GMT Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 252 v=0 o=CiscoSystemsSIP-GW-UserAgent 5682 8981 IN IP4 w.x.y.z s=SIP Call c=IN IP4 w.x.y.z t=0 0 m=audio 17596 RTP/AVP 4 101 c=IN IP4 w.x.y.z a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 14 headers, 11 lines Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port w.x.y.z:17596 Found description format G723 Found description format telephone-event Capabilities: us - 0x1 (g723), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) !!!!!!!!!!! According to tcpdump RTP stream from w.x.y.z:17596 starts here !!!!!!!!!!! Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK5852b69f From: "2071234567" ;tag=as4e6afb66 To: ;tag=568D5E5C-1ACD Date: Fri, 04 Mar 2005 17:34:08 GMT Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 252 v=0 o=CiscoSystemsSIP-GW-UserAgent 5682 8981 IN IP4 w.x.y.z s=SIP Call c=IN IP4 w.x.y.z t=0 0 m=audio 17596 RTP/AVP 4 101 c=IN IP4 w.x.y.z a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 14 headers, 11 lines Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port w.x.y.z:17596 Found description format G723 Found description format telephone-event Capabilities: us - 0x1 (g723), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to a.b.c.e, port 5060 Transmitting: ACK sip:14151234567@w.x.y.z:5060 SIP/2.0 Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK2a4f1588 Route: From: "2071234567" ;tag=as4e6afb66 To: ;tag=568D5E5C-1ACD Contact: Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to a.b.c.e:5060 -- Executing Answer("SIP/p.r.s.t-895d", "") in new stack -- Executing AbsoluteTimeout("SIP/p.r.s.t-895d", "90") in new stack -- Set Absolute Timeout to 90 -- Executing Wait("SIP/p.r.s.t-895d", "10") in new stack -- Executing WaitForSilence("SIP/p.r.s.t-895d", "2000") in new stack -- Waiting 1 time(s) for 2000 ms silence Mar 4 17:34:31 WARNING[23130]: app_waitforsilence.c:76 do_waiting: One waitfor failed, trying another Mar 4 17:34:33 WARNING[23130]: app_waitforsilence.c:80 do_waiting: No audio available on SIP/p.r.s.t-895d?? -- Executing Playback("SIP/p.r.s.t-895d", "/root/sip/karyn_test") in new stack !!!!!!!!!!! INTERESTING PART HERE !!!!!!!!!!! G723 opened: fd=16, name=0x403dd7f0, base=0x8125540 Sent RTP packet to w.x.y.z:17596 (type 4, seq 55200, ts 240, len 24) -- Playing '/root/sip/karyn_test' (language 'en') !!!!!!!!!!! First packet seen by Asterisk !!!!!!!!!!! Got RTP packet from w.x.y.z:17596 (type 4, seq 4412, ts 1847636558, len 24) Sent RTP packet to w.x.y.z:17596 (type 4, seq 55201, ts 480, len 24) Got RTP packet from w.x.y.z:17596 (type 4, seq 4413, ts 1847636798, len 24) Sent RTP packet to w.x.y.z:17596 (type 4, seq 55202, ts 720, len 24) Got RTP packet from w.x.y.z:17596 (type 4, seq 4414, ts 1847637038, len 24) <...skip...> g723 asked to close 16, 0x403dd7f0, base=0x8125540 -- Executing Hangup("SIP/p.r.s.t-895d", "") in new stack == Spawn extension (test, 4151234567, 6) exited non-zero on 'SIP/p.r.s.t-895d' set_destination: Parsing for address/port to send to set_destination: set destination to a.b.c.e, port 5060 Reliably Transmitting: BYE sip:14151234567@w.x.y.z:5060 SIP/2.0 Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK0e0cbdd0 Route: From: "2071234567" ;tag=as4e6afb66 To: ;tag=568D5E5C-1ACD Contact: Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to a.b.c.e:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK0e0cbdd0 From: "2071234567" ;tag=as4e6afb66 To: ;tag=568D5E5C-1ACD Date: Fri, 04 Mar 2005 17:35:08 GMT Call-ID: 7c83e458257130a362bbd95a436c6125@a.b.c.d Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE 9 headers, 0 lines Destroying call '7c83e458257130a362bbd95a436c6125@a.b.c.d'