asterisk*CLI> set verbose 4 Verbosity was 3 and is now 4 asterisk*CLI> set debug 4 Core debug was 0 and is now 4 asterisk*CLI> sip debug SIP Debugging enabled -- Accepting AUTHENTICATED call from 24.2XX.X.XX: > requested format = ulaw, > requested prefs = (ulaw), > actual format = ulaw, > host prefs = (ulaw), > priority = mine -- Executing Set("IAX2/321-4", "_ALERT_INFO=info=alert-autoanswer") in new stack -- Executing Page("IAX2/321-4", "SIP/211&SIP/212") in new stack -- Playing 'beep' (language 'fr') We're at 192.168.44.13 port 10060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.44.72:5060: INVITE sip:211@192.168.44.72 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK758404ba;rport From: "David" ;tag=as10fa6c8f To: Contact: Call-ID: 001cc96205b2313f2e45a1df12561140@192.168.44.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Jun 2006 04:06:29 GMT Alert-Info: info=alert-autoanswer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 4262 4262 IN IP4 192.168.44.13 s=session c=IN IP4 192.168.44.13 t=0 0 m=audio 10060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- We're at 192.168.44.13 port 10088 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.44.53:5060: INVITE sip:212@192.168.44.53 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK43de9afa;rport From: "David" ;tag=as2463276e To: Contact: Call-ID: 5089bd1d330e00042b6c393d28fbe6e5@192.168.44.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Jun 2006 04:06:29 GMT Alert-Info: info=alert-autoanswer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 4262 4262 IN IP4 192.168.44.13 s=session c=IN IP4 192.168.44.13 t=0 0 m=audio 10088 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> <-- SIP read from 192.168.44.53:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK43de9afa;rport From: "David" ;tag=as2463276e To: ;tag=65CC9D4C-1ABF1C83 CSeq: 102 INVITE Call-ID: 5089bd1d330e00042b6c393d28fbe6e5@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 --- (9 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.44.72:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK758404ba;rport From: "David" ;tag=as10fa6c8f To: ;tag=4C6A5409-B781C24C CSeq: 102 INVITE Call-ID: 001cc96205b2313f2e45a1df12561140@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 --- (9 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.44.53:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK43de9afa;rport From: "David" ;tag=as2463276e To: ;tag=65CC9D4C-1ABF1C83 CSeq: 102 INVITE Call-ID: 5089bd1d330e00042b6c393d28fbe6e5@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.44.72:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK758404ba;rport From: "David" ;tag=as10fa6c8f To: ;tag=4C6A5409-B781C24C CSeq: 102 INVITE Call-ID: 001cc96205b2313f2e45a1df12561140@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- -- Hungup 'IAX2/321-4' asterisk*CLI> <-- SIP read from 192.168.44.53:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK43de9afa;rport From: "David" ;tag=as2463276e To: ;tag=65CC9D4C-1ABF1C83 CSeq: 102 INVITE Call-ID: 5089bd1d330e00042b6c393d28fbe6e5@192.168.44.13 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1149653256 1149653256 IN IP4 192.168.44.53 s=Polycom IP Phone c=IN IP4 192.168.44.53 t=0 0 m=audio 2264 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.44.53:2264 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.44.53, port 5060 Transmitting (no NAT) to 192.168.44.53:5060: ACK sip:212@192.168.44.53 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK07f3eadd;rport From: "David" ;tag=as2463276e To: ;tag=65CC9D4C-1ABF1C83 Contact: Call-ID: 5089bd1d330e00042b6c393d28fbe6e5@192.168.44.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Launching MeetMe(735814361d|mqxdw) on SIP/212-5c9e -- Created MeetMe conference 1023 for conference '735814361d' asterisk*CLI> <-- SIP read from 192.168.44.72:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK758404ba;rport From: "David" ;tag=as10fa6c8f To: ;tag=4C6A5409-B781C24C CSeq: 102 INVITE Call-ID: 001cc96205b2313f2e45a1df12561140@192.168.44.13 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1149653256 1149653256 IN IP4 192.168.44.72 s=Polycom IP Phone c=IN IP4 192.168.44.72 t=0 0 m=audio 2240 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.44.72:2240 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.44.72, port 5060 Transmitting (no NAT) to 192.168.44.72:5060: ACK sip:211@192.168.44.72 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK57c54454;rport From: "David" ;tag=as10fa6c8f To: ;tag=4C6A5409-B781C24C Contact: Call-ID: 001cc96205b2313f2e45a1df12561140@192.168.44.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Launching MeetMe(735814361d|mqxdw) on SIP/211-d60c asterisk*CLI> show channels Channel Location State Application(Data) Zap/pseudo-146131923 s@internal:1 Rsrvd (None) SIP/212-5c9e (None) Up MeetMe(735814361d|mqxdw) SIP/211-d60c (None) Up MeetMe(735814361d|mqxdw) 3 active channels 0 active calls