asterisk:~# asterisk -r Asterisk 1.0.5-BRIstuffed-0.2.0-RC7i, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk 1.0.5-BRIstuffed-0.2.0-RC7i currently running on asterisk (pid = 28717) Verbosity is at least 4 asterisk*CLI> sip debug peer 350 SIP Debugging Enabled for IP: 10.10.10.19:2072 asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> -- Executing Dial("SIP/345-5f9e", "Agent/350") in new stack -- outgoing agentcall, to agent '350', on 'Local/350@oletus-44ab,1' -- Called 350 -- Executing Macro("Local/350@oletus-44ab,2", "sippuh|350|45|350") in new stack -- Executing DBget("Local/350@oletus-44ab,2", "temp=CFIM/350") in new stack -- DBget: varname=temp, family=CFIM, key=350 -- DBget: Value not found in database. -- Executing Dial("Local/350@oletus-44ab,2", "SIP/350|45|Ttr") in new stack We're at 80.95.135.47 port 18534 Video is at 80.95.135.47 port 10602 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Answering with capability 0x10 (g726) Answering with capability 0x20 (adpcm) Answering with capability 0x40 (slin) Answering with capability 0x80 (lpc10) Answering with capability 0x100 (g729) Answering with capability 0x200 (speex) Answering with capability 0x400 (ilbc) Answering with capability 0x10000 (jpeg) Answering with capability 0x20000 (png) Answering with capability 0x40000 (h261) Answering with capability 0x80000 (h263) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 23 lines Reliably Transmitting: INVITE sip:350@10.10.10.19:2072;line=o8yvl07l SIP/2.0 Via: SIP/2.0/UDP xx.xx.xxx.xx:5060;branch=z9hG4bK478bb0c6 From: "Jukka analog" ;tag=as20f61c61 To: Contact: Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 102 INVITE User-Agent: Netland Asterisk Date: Thu, 07 Apr 2005 06:33:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 547 v=0 o=root 28781 28781 IN IP4 xx.xx.xxx.xx s=session c=IN IP4 xx.xx.xxx.xx t=0 0 m=audio 18534 RTP/AVP 0 3 8 2 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 10602 RTP/AVP 26 31 34 a=rtpmap:26 JPEG/90000 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 (no NAT) to 10.10.10.19:2072 -- Called 350 -- Agent/350 is ringing asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP xx.xx.xxx.xx:5060;branch=z9hG4bK478bb0c6 From: "Jukka analog" ;tag=as20f61c61 To: ;tag=rzri5c78eq Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 10 headers, 0 lines -- SIP/350-585e is ringing asterisk*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP xx.xx.xxx.xx:5060;branch=z9hG4bK478bb0c6 From: "Jukka analog" ;tag=as20f61c61 To: ;tag=rzri5c78eq Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 10 headers, 0 lines -- SIP/350-585e is ringing asterisk*CLI> Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP xx.xx.xxx.xx:5060;branch=z9hG4bK478bb0c6 From: "Jukka analog" ;tag=as20f61c61 To: ;tag=rzri5c78eq Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 102 INVITE Contact: User-Agent: snom360-3.57r Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Type: application/sdp Content-Length: 250 v=0 o=root 457584503 457584503 IN IP4 10.10.10.19 s=call c=IN IP4 10.10.10.19 t=0 0 m=audio 61902 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 0.9 : user 9kksj== 10.10.10.19 61902 a=sendrecv 13 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.19:61902 Found description format pcmu Found description format telephone-event Capabilities: us - 0xf07fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.19, port 2072 Transmitting: ACK sip:350@10.10.10.19:2072;line=o8yvl07l SIP/2.0 Via: SIP/2.0/UDP xx.xx.xxx.xx:5060;branch=z9hG4bK118e9220 From: "Jukka analog" ;tag=as20f61c61 To: ;tag=rzri5c78eq Contact: Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 102 ACK User-Agent: Netland Asterisk Content-Length: 0 (no NAT) to 10.10.10.19:2072 -- SIP/350-585e answered Local/350@oletus-44ab,2 -- Agent/350 stopped sounds -- Agent/350 answered SIP/345-5f9e Apr 7 09:33:18 WARNING[28780]: channel.c:515 ast_channel_walk_locked: Avoided deadlock for 'Local/350@oletus-44ab,2', 10 retries! == Spawn extension (macro-sippuh, s, 102) exited non-zero on 'Local/350@oletus-44ab,2' in macro 'sippuh' == Spawn extension (oletus, 350, 1) exited non-zero on 'Local/350@oletus-44ab,2' asterisk*CLI> Sip read: INVITE sip:345@xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-yttvl80h0oxm;rport From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 1 INVITE Max-Forwards: 70 Contact: P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360-3.57r Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 336 v=0 o=root 457584503 457584504 IN IP4 10.10.10.19 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 61902 RTP/AVP 0 101 k=base64:7R2qlMpGQqqNZEs5W5ebWHVRy0d12EhMSdg+54eyMao= a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 10.10.10.19 61902 a=sendonly 18 headers, 14 lines Using latest request as basis request Sending to 10.10.10.19 : 2072 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:61902 Found description format pcmu Found description format telephone-event Capabilities: us - 0xf07fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at xx.xx.xxx.xx port 18534 Video is at xx.xx.xxx.xx port 10602 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Answering with capability 0x10 (g726) Answering with capability 0x20 (adpcm) Answering with capability 0x40 (slin) Answering with capability 0x80 (lpc10) Answering with capability 0x100 (g729) Answering with capability 0x200 (speex) Answering with capability 0x400 (ilbc) Answering with capability 0x10000 (jpeg) Answering with capability 0x20000 (png) Answering with capability 0x40000 (h261) Answering with capability 0x80000 (h263) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-yttvl80h0oxm From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 1 INVITE User-Agent: Netland Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 443 v=0 o=root 28781 28782 IN IP4 xx.xx.xxx.xx s=session c=IN IP4 xx.xx.xxx.xx t=0 0 m=audio 18534 RTP/AVP 0 3 8 2 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.10.10.19:2072 asterisk*CLI> Sip read: ACK sip:345@xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-dqjq0pbqc7m9;rport From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 1 ACK Max-Forwards: 70 Contact: Content-Length: 0 9 headers, 0 lines asterisk*CLI> Sip read: INVITE sip:345@xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-xx533mamz31k;rport From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 2 INVITE Max-Forwards: 70 Contact: P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360-3.57r Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 340 v=0 o=root 457584503 457584505 IN IP4 10.10.10.19 s=call c=IN IP4 10.10.10.19 t=0 0 m=audio 61902 RTP/AVP 0 101 k=base64:7R2qlMpGQqqNZEs5W5ebWHVRy0d12EhMSdg+54eyMao= a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=encryption:optional a=alt:1 0.9 : user 9kksj== 10.10.10.19 61902 a=sendrecv 18 headers, 14 lines Using latest request as basis request Sending to 10.10.10.19 : 2072 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.19:61902 Found description format pcmu Found description format telephone-event Capabilities: us - 0xf07fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at xx.xx.xxx.xx port 18534 Video is at xx.xx.xxx.xx port 10602 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Answering with capability 0x10 (g726) Answering with capability 0x20 (adpcm) Answering with capability 0x40 (slin) Answering with capability 0x80 (lpc10) Answering with capability 0x100 (g729) Answering with capability 0x200 (speex) Answering with capability 0x400 (ilbc) Answering with capability 0x10000 (jpeg) Answering with capability 0x20000 (png) Answering with capability 0x40000 (h261) Answering with capability 0x80000 (h263) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-xx533mamz31k From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 2 INVITE User-Agent: Netland Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 443 v=0 o=root 28781 28783 IN IP4 xx.xx.xxx.xx s=session c=IN IP4 xx.xx.xxx.xx t=0 0 m=audio 18534 RTP/AVP 0 3 8 2 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.10.10.19:2072 asterisk*CLI> Sip read: ACK sip:345@xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-blmw9ji6tlht;rport From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 2 ACK Max-Forwards: 70 Contact: Content-Length: 0 9 headers, 0 lines asterisk*CLI> Sip read: BYE sip:345@xx.xx.xxx.xx SIP/2.0 Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-as37jp0g65ka;rport From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 3 BYE Max-Forwards: 70 Contact: User-Agent: snom360-3.57r Content-Length: 0 10 headers, 0 lines Sending to 10.10.10.19 : 2072 (non-NAT) Transmitting (no NAT): IP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.19:2072;branch=z9hG4bK-as37jp0g65ka From: ;tag=rzri5c78eq To: "Jukka analog" ;tag=as20f61c61 Call-ID: 7c344efa234363113226b02207e67564@xx.xx.xxx.xx CSeq: 3 BYE User-Agent: Netland Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.10.10.19:2072 == Spawn extension (oletus, 998, 1) exited non-zero on 'SIP/345-5f9e' Destroying call '7c344efa234363113226b02207e67564@xx.xx.xxx.xx' asterisk*CLI>