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 = 18231) phone*CLI> -- Remote UNIX connection Verbosity is at least 9 phone*CLI> Destroying call '3deae60e1f50a9472b833daf6114e2eb@.com' phone*CLI> Sep 9 17:39:37 NOTICE[18232]: 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> REGISTER attempt 1 to (VONAGE_USERNAME)@sphone.vopr.vonage.net 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=z9hG4bK7e97a45fFrom: ;tag=as7c2b5ffeTo: Call-ID: 3deae60e1f50a9472b833daf6114e2eb@.comCSeq: 148 REGISTERUser-Agent: PhoneExpires: 120Contact: Event: registrationContent-Length: 0 --- phone*CLI> -- Executing SetGroup("SIP/370-f868", "vonage3") in new stack -- Executing CheckGroup("SIP/370-f868", "1") in new stack -- Executing Dial("SIP/370-f868", "SIP/(NUMBER_CALLED)@vonage3") in new stack We're at (ASTERISK_IP) port 11528 Video is at (ASTERISK_IP) port 18862 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 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=z9hG4bK45f6b83bFrom: "Adrian" ;tag=as7f8ee5f1To: Contact: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 102 INVITEUser-Agent: PhoneDate: Sat, 10 Sep 2005 00:39:37 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 431v=0o=root 18232 18232 IN IP4 (ASTERISK_IP)s=sessionc=IN IP4 (ASTERISK_IP)t=0 0m=audio 11528 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 18862 RTP/AVP 31 34 103a=rtpmap:31 H261/90000a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000 --- phone*CLI> -- Called (NUMBER_CALLED)@vonage3 phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OKVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK7e97a45fFrom: ;tag=as7c2b5ffeTo: Call-ID: 3deae60e1f50a9472b833daf6114e2eb@.comCSeq: 148 REGISTERContact: ;expires=20Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Scheduling destruction of call '3deae60e1f50a9472b833daf6114e2eb@.com' in 32000 ms phone*CLI> Sep 9 17:39:37 NOTICE[18232]: 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/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK45f6b83bFrom: "Adrian" ;tag=as7f8ee5f1To: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 102 INVITEProxy-Authenticate: Digest realm="216.115.25.198", domain="sip:216.115.25.198", nonce="2064171456", algorithm=MD5Max-Forwards: 15Content-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=z9hG4bK45f6b83bFrom: "Adrian" ;tag=as7f8ee5f1To: Contact: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 102 ACKUser-Agent: PhoneContent-Length: 0 --- phone*CLI> We're at (ASTERISK_IP) port 11528 phone*CLI> Video is at (ASTERISK_IP) port 18862 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) phone*CLI> Answering with preferred capability 0x80000 (h263) Answering with preferred capability 0x100000 (h263p) phone*CLI> 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=z9hG4bK2afa836cFrom: "Adrian" ;tag=as7f8ee5f1To: Contact: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEUser-Agent: 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 00:39:37 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 431v=0o=root 18232 18233 IN IP4 (ASTERISK_IP)s=sessionc=IN IP4 (ASTERISK_IP)t=0 0m=audio 11528 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 18862 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 TryingVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2afa836cFrom: "Adrian" ;tag=as7f8ee5f1To: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEMax-Forwards: 15Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 180 RingingVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2afa836cFrom: "Adrian" ;tag=as7f8ee5f1To: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEMax-Forwards: 15Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> -- SIP/vonage3-a6a4 is ringing phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2afa836cFrom: "Adrian" ;tag=as7f8ee5f1To: ;tag=2067764114Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEContact: ;rtpupdated=-Max-Forwards: 15Content-Type: application/sdpContent-Length: 218v=0o=- 740958 0 IN IP4 64.192.218.178s=Cisco SDP 0c=IN IP4 64.192.218.178t=0 0m=audio 20636 RTP/AVP 0 101 100a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194 phone*CLI> --- (10 headers 10 lines)--- phone*CLI> Found RTP audio format 0 phone*CLI> Found RTP audio format 101 Found RTP audio format 100 phone*CLI> Peer audio RTP is at port 64.192.218.178:20636 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-a6a4 is making progress passing it to SIP/370-f868 phone*CLI> Sep 9 17:39:42 NOTICE[18232]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 69.28.248.106 phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2afa836cFrom: "Adrian" ;tag=as7f8ee5f1To: ;tag=2067764114Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEContact: ;rtpupdated=-Max-Forwards: 15Content-Type: application/sdpContent-Length: 218v=0o=- 740958 0 IN IP4 64.192.218.178s=Cisco SDP 0c=IN IP4 64.192.218.178t=0 0m=audio 20636 RTP/AVP 0 101 100a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194 phone*CLI> --- (10 headers 10 lines)--- phone*CLI> Found RTP audio format 0 phone*CLI> Found RTP audio format 101 Found RTP audio format 100 phone*CLI> Peer audio RTP is at port 64.192.218.178:20636 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-a6a4 is making progress passing it to SIP/370-f868 phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2afa836cFrom: "Adrian" ;tag=as7f8ee5f1To: ;tag=2067764114Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEContact: ;rtpupdated=-Max-Forwards: 15Content-Type: application/sdpContent-Length: 218v=0o=- 740958 0 IN IP4 64.192.218.178s=Cisco SDP 0c=IN IP4 64.192.218.178t=0 0m=audio 20636 RTP/AVP 0 101 100a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194 phone*CLI> --- (10 headers 10 lines)--- phone*CLI> Found RTP audio format 0 phone*CLI> Found RTP audio format 101 Found RTP audio format 100 phone*CLI> Peer audio RTP is at port 64.192.218.178:20636 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-a6a4 is making progress passing it to SIP/370-f868 phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OKVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2afa836cRecord-Route: From: "Adrian" ;tag=as7f8ee5f1To: ;tag=1226097548Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 INVITEContact: ;rtpupdated=-Max-Forwards: 15Content-Type: application/sdpContent-Length: 218v=0o=- 740958 0 IN IP4 64.192.218.178s=Cisco SDP 0c=IN IP4 64.192.218.178t=0 0m=audio 20636 RTP/AVP 0 101 100a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=rtpmap:100 X-NSE/8000a=fmtp:100 192-194 phone*CLI> --- (11 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 64.192.218.178:20636 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> Transmitting (no NAT) to 216.115.25.198:5061: ACK sip:(NUMBER_CALLED)@216.115.25.198:5061 SIP/2.0Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK3a91633eRoute: From: "Adrian" ;tag=as7f8ee5f1To: ;tag=1226097548Contact: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 103 ACKUser-Agent: PhoneContent-Length: 0 phone*CLI> --- phone*CLI> -- SIP/vonage3-a6a4 answered SIP/370-f868 phone*CLI> -- Attempting native bridge of SIP/370-f868 and SIP/vonage3-a6a4 phone*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 216.115.25.198, port 5061 Reliably Transmitting (no NAT) to 216.115.25.198:5061: BYE sip:(NUMBER_CALLED)@216.115.25.198:5061 SIP/2.0Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK376713bcRoute: From: "Adrian" ;tag=as7f8ee5f1To: ;tag=1226097548Contact: Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 104 BYEUser-Agent: PhoneProxy-Authorization: Digest username="(VONAGE_USERNAME)", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="2064171456", response="a4e5cc7322581c92d664a4c3aadecd46", opaque=""Content-Length: 0 --- phone*CLI> <-- SIP read from 216.115.25.198:5061: SIP/2.0 200 OKVia: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK376713bcFrom: "Adrian" ;tag=as7f8ee5f1To: ;tag=1226097548Call-ID: 0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.netCSeq: 104 BYEMax-Forwards: 15Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Destroying call '0e423fc556186dd66d0e3412499d6f8d@sphone.vopr.vonage.net' phone*CLI> Destroying call '3deae60e1f50a9472b833daf6114e2eb@.com' phone*CLI> Sep 9 17:39:53 NOTICE[18232]: 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> REGISTER attempt 1 to (VONAGE_USERNAME)@sphone.vopr.vonage.net 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=z9hG4bK29b9c3b8From: ;tag=as5d28e953To: Call-ID: 3deae60e1f50a9472b833daf6114e2eb@.comCSeq: 149 REGISTERUser-Agent: PhoneExpires: 120Contact: Event: registrationContent-Length: 0 --- phone*CLI>