Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Asterisk CVS HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS HEAD currently running on phone (pid = 18296) phone*CLI> Verbosity was 8 and is now 9 phone*CLI> -- Executing SetGroup("SIP/370-c87c", "vonage3") in new stack Sep 9 17:57:25 WARNING[18297]: app_groupcount.c:110 group_set_exec: The SetGroup application has been deprecated, please use the GROUP() function. -- Executing CheckGroup("SIP/370-c87c", "1") in new stack Sep 9 17:57:25 WARNING[18297]: app_groupcount.c:135 group_check_exec: The CheckGroup application has been deprecated, please use a combination of the GotoIf application and the GROUP_COUNT() function. -- Executing Dial("SIP/370-c87c", "SIP/(NUMBER_CALLED)@vonage3") in new stack We're at (ASTERISK_IP) port 19964 Video is at (ASTERISK_IP) port 11696 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x200 (speex) Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x40000 (h261) Answering with preferred capability 0x80000 (h263) Answering with preferred capability 0x100000 (h263p) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 18 lines Reliably Transmitting (no NAT) to 216.115.25.198:5061: INVITE sip:(NUMBER_CALLED)@sphone.vopr.vonage.net:5061 SIP/2.0Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK677957f3From: "Adrian" ;tag=as640fbb8eTo: Contact: Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.netCSeq: 102 INVITEUser-Agent: (DOMAIN) PhoneDate: Sat, 10 Sep 2005 00:57:25 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 431v=0o=root 18297 18297 IN IP4 (ASTERISK_IP)s=sessionc=IN IP4 (ASTERISK_IP)t=0 0m=audio 19964 RTP/AVP 0 8 3 110 97 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:110 speex/8000a=rtpmap:97 iLBC/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -m=video 11696 RTP/AVP 31 34 103a=rtpmap:31 H261/90000a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000 --- -- Called (NUMBER_CALLED)@vonage3 phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK677957f3 From: "Adrian" ;tag=as640fbb8e To: Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 102 INVITE Proxy-Authenticate: Digest realm="216.115.25.198", domain="sip:216.115.25.198", nonce="2064171456", algorithm=MD5 Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (9 headers 0 lines)--- phone*CLI> Transmitting (no NAT) to 216.115.25.198:5061: ACK sip:(NUMBER_CALLED)@sphone.vopr.vonage.net:5061 SIP/2.0Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK677957f3From: "Adrian" ;tag=as640fbb8eTo: Contact: Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.netCSeq: 102 ACKUser-Agent: (DOMAIN) PhoneContent-Length: 0 --- phone*CLI> We're at (ASTERISK_IP) port 19964 phone*CLI> Video is at (ASTERISK_IP) port 11696 phone*CLI> Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) phone*CLI> Answering with preferred capability 0x2 (gsm) phone*CLI> Answering with preferred capability 0x200 (speex) Answering with preferred capability 0x400 (ilbc) phone*CLI> Answering with preferred capability 0x40000 (h261) Answering with preferred capability 0x80000 (h263) phone*CLI> Answering with preferred capability 0x100000 (h263p) Answering with non-codec capability 0x1 (telephone-event) phone*CLI> Reliably Transmitting (no NAT) to 216.115.25.198:5061: INVITE sip:(NUMBER_CALLED)@sphone.vopr.vonage.net:5061 SIP/2.0Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1fFrom: "Adrian" ;tag=as640fbb8eTo: Contact: Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.netCSeq: 103 INVITEUser-Agent: (DOMAIN) PhoneProxy-Authorization: Digest username="(VONAGE_USERNAME)", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="2064171456", response="57dd68d77c52c67512b8ec482ab5fe39", opaque=""Date: Sat, 10 Sep 2005 00:57:25 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 431v=0o=root 18297 18298 IN IP4 (ASTERISK_IP)s=sessionc=IN IP4 (ASTERISK_IP)t=0 0m=audio 19964 RTP/AVP 0 8 3 110 97 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:110 speex/8000a=rtpmap:97 iLBC/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -m=video 11696 RTP/AVP 31 34 103a=rtpmap:31 H261/90000a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000 --- phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f From: "Adrian" ;tag=as640fbb8e To: Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Destroying call '111e5ec00713efe9180a641e71dd623c@(DOMAIN).com' phone*CLI> Sep 9 17:57:29 NOTICE[18297]: chan_sip.c:4955 sip_reregister: -- Re-registration for (VONAGE_USERNAME)@sphone.vopr.vonage.net phone*CLI> -- parse_srv: SRV mapped to host sphone.vopr.vonage.net, port 5061 phone*CLI> REGISTER 11 headers, 0 lines phone*CLI> Reliably Transmitting (no NAT) to 216.115.25.198:5061: REGISTER sip:sphone.vopr.vonage.net SIP/2.0Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK4e23c2c2From: ;tag=as5eda4394To: Call-ID: 111e5ec00713efe9180a641e71dd623c@(DOMAIN).comCSeq: 104 REGISTERUser-Agent: (DOMAIN) PhoneExpires: 120Contact: Event: registrationContent-Length: 0 --- phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK4e23c2c2 From: ;tag=as5eda4394 To: Call-ID: 111e5ec00713efe9180a641e71dd623c@(DOMAIN).com CSeq: 104 REGISTER Contact: ;expires=20 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Scheduling destruction of call '111e5ec00713efe9180a641e71dd623c@(DOMAIN).com' in 32000 ms phone*CLI> Sep 9 17:57:29 NOTICE[18297]: chan_sip.c:9178 handle_response_register: Outbound Registration: Expiry for sphone.vopr.vonage.net is 20 sec (Scheduling reregistration in 15 s) phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 180 Ringing Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f From: "Adrian" ;tag=as640fbb8e To: Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> -- SIP/vonage3-fe9d is ringing phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f From: "Adrian" ;tag=as640fbb8e To: ;tag=2067764114 Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Contact: ;rtpupdated=- Max-Forwards: 15 Content-Type: application/sdp Content-Length: 218 v=0 o=- 759592 0 IN IP4 64.192.218.178 s=Cisco SDP 0 c=IN IP4 64.192.218.178 t=0 0 m=audio 17870 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 phone*CLI> --- (10 headers 10 lines)--- phone*CLI> Found RTP audio format 0 phone*CLI> Found RTP audio format 101 phone*CLI> Found RTP audio format 100 Peer audio RTP is at port 64.192.218.178:17870 phone*CLI> Peer video RTP is at port 64.192.218.178:65535 phone*CLI> Found description format telephone-event phone*CLI> Found description format X-NSE phone*CLI> Capabilities: us - 0x1c060e (gsm|ulaw|alaw|speex|ilbc|h261|h263|h263p), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) phone*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) phone*CLI> -- SIP/vonage3-fe9d is making progress passing it to SIP/370-c87c phone*CLI> Sep 9 17:57:29 NOTICE[18297]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: (CLIENT IP) phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f Record-Route: From: "Adrian" ;tag=as640fbb8e To: ;tag=450497376 Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Contact: ;rtpupdated=- Max-Forwards: 15 Content-Type: application/sdp Content-Length: 218 phone*CLI> v=0 o=- 759592 0 IN IP4 64.192.218.178 s=Cisco SDP 0 c=IN IP4 64.192.218.178 t=0 0 m=audio 17870 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 phone*CLI> --- (11 headers 10 lines)--- phone*CLI> Destroying call '21bf444537e69e9547101670727174df@sphone.vopr.vonage.net' phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f Record-Route: From: "Adrian" ;tag=as640fbb8e To: ;tag=450497376 Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Contact: ;rtpupdated=- Max-Forwards: 15 Content-Type: application/sdp Content-Length: 218 phone*CLI> v=0 o=- 759592 0 IN IP4 64.192.218.178 s=Cisco SDP 0 c=IN IP4 64.192.218.178 t=0 0 m=audio 17870 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 phone*CLI> --- (11 headers 10 lines)--- phone*CLI> Destroying call '21bf444537e69e9547101670727174df@sphone.vopr.vonage.net' phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f Record-Route: From: "Adrian" ;tag=as640fbb8e To: ;tag=450497376 Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Contact: ;rtpupdated=- Max-Forwards: 15 Content-Type: application/sdp Content-Length: 218 phone*CLI> v=0 o=- 759592 0 IN IP4 64.192.218.178 s=Cisco SDP 0 c=IN IP4 64.192.218.178 t=0 0 m=audio 17870 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 phone*CLI> --- (11 headers 10 lines)--- phone*CLI> Destroying call '21bf444537e69e9547101670727174df@sphone.vopr.vonage.net' phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK456a7d1f Record-Route: From: "Adrian" ;tag=as640fbb8e To: ;tag=450497376 Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 103 INVITE Contact: ;rtpupdated=- Max-Forwards: 15 Content-Type: application/sdp Content-Length: 218 phone*CLI> v=0 o=- 759592 0 IN IP4 64.192.218.178 s=Cisco SDP 0 c=IN IP4 64.192.218.178 t=0 0 m=audio 17870 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 phone*CLI> --- (11 headers 10 lines)--- phone*CLI> Destroying call '21bf444537e69e9547101670727174df@sphone.vopr.vonage.net' phone*CLI> <-- SIP read from 216.115.25.198:5061: BYE sip:(VONAGE_USERNAME)@(ASTERISK_IP):5070 SIP/2.0 Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.171:5060;branch=orig-8129a8-(VONAGE_USERNAME)-(NUMBER_CALLED) From: ;tag=450497376 To: "Adrian" ;tag=as640fbb8e Call-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.net CSeq: 1 BYE Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (9 headers 0 lines)--- phone*CLI> Transmitting (no NAT) to 216.115.25.198:5061: SIP/2.0 481 Call/Transaction Does Not ExistVia: SIP/2.0/UDP 216.115.25.198:5061Via: SIP/2.0/UDP 216.115.20.171:5060;branch=orig-8129a8-(VONAGE_USERNAME)-(NUMBER_CALLED)From: ;tag=450497376To: "Adrian" ;tag=as640fbb8eCall-ID: 21bf444537e69e9547101670727174df@sphone.vopr.vonage.netCSeq: 1 BYEUser-Agent: (DOMAIN) PhoneAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0 --- phone*CLI> Destroying call '21bf444537e69e9547101670727174df@sphone.vopr.vonage.net' phone*CLI>