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 = 19115) phone*CLI> Verbosity is at least 6 -- Remote UNIX connection phone*CLI> Destroying call '54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com' phone*CLI> Sep 9 21:57:33 NOTICE[19116]: chan_sip.c:4955 sip_reregister: -- Re-registration for (ACCOUNT1)@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/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK078cfe07 From: ;tag=as6c8038b7 To: Call-ID: 54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com CSeq: 132 REGISTER User-Agent: (MYDOMAIN) Phone Expires: 120 Contact: Event: registration Content-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=z9hG4bK078cfe07 From: ;tag=as6c8038b7 To: Call-ID: 54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com CSeq: 132 REGISTER Contact: ;expires=20 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Scheduling destruction of call '54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com' in 32000 ms phone*CLI> Sep 9 21:57:33 NOTICE[19116]: 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: INVITE sip:(ACCOUNT1)@(ASTERISK_IP):5070;suppress-features=- SIP/2.0 Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bK15BD Record-Route: Record-Route: From: ;tag=781947608 To: Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 101 INVITE Contact: ;rtpupdated=- Max-Forwards: 13 Content-Type: application/sdp Content-Length: 361 v=0 o=CiscoSystemsSIP-GW-UserAgent 5215 5825 IN IP4 216.18.39.139 s=SIP Call c=IN IP4 216.18.39.139 t=0 0 m=audio 16664 RTP/AVP 0 18 2 100 101 c=IN IP4 216.18.39.139 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 phone*CLI> --- (14 headers 15 lines)--- phone*CLI> Using INVITE request as basis request - 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 phone*CLI> Sending to 216.115.25.198 : 5061 (non-NAT) phone*CLI> Found peer 'vonage7' phone*CLI> Found RTP audio format 0 phone*CLI> Found RTP audio format 18 Found RTP audio format 2 phone*CLI> Found RTP audio format 100 phone*CLI> Found RTP audio format 101 Peer audio RTP is at port 216.18.39.139:16664 phone*CLI> Peer video RTP is at port 216.18.39.139:65535 phone*CLI> Found description format PCMU phone*CLI> Found description format G729 phone*CLI> Found description format G726-32 Found description format X-NSE phone*CLI> Found description format telephone-event phone*CLI> Capabilities: us - 0x1c060e (gsm|ulaw|alaw|speex|ilbc|h261|h263|h263p), peer - audio=0x114 (ulaw|g726|g729)/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> Looking for (ACCOUNT1) in vonage_incoming phone*CLI> list_route: hop: phone*CLI> list_route: hop: list_route: hop: phone*CLI> Transmitting (no NAT) to 216.115.25.198:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bK15BD From: ;tag=781947608 To: Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 101 INVITE User-Agent: (MYDOMAIN) Phone Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: C phone*CLI> ontent-Length: 0 --- phone*CLI> -- Executing Goto("SIP/(ACCOUNT2)-5202", "incoming|s|1") in new stack -- Goto (incoming,s,1) -- Executing Ringing("SIP/(ACCOUNT2)-5202", "") in new stack phone*CLI> Transmitting (no NAT) to 216.115.25.198:5061: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bK15BD From: ;tag=781947608 To: ;tag=as36c4b92b Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 101 INVITE User-Agent: (MYDOMAIN) Phone Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- phone*CLI> -- Executing Wait("SIP/(ACCOUNT2)-5202", "10") in new stack phone*CLI> Destroying call '33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com' phone*CLI> Sep 9 21:57:34 NOTICE[19116]: chan_sip.c:4955 sip_reregister: -- Re-registration for (ACCOUNT2)@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/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK60672b36 From: ;tag=as51102c28 To: Call-ID: 33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com CSeq: 132 REGISTER User-Agent: (MYDOMAIN) Phone Expires: 120 Contact: Event: registration Content-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=z9hG4bK60672b36 From: ;tag=as51102c28 To: Call-ID: 33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com CSeq: 132 REGISTER Contact: ;expires=20 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Scheduling destruction of call '33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com' in 32000 ms phone*CLI> Sep 9 21:57:34 NOTICE[19116]: 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> -- Executing NoOp("SIP/(ACCOUNT2)-5202", "") in new stack phone*CLI> -- Executing Answer("SIP/(ACCOUNT2)-5202", "") in new stack phone*CLI> We're at (ASTERISK_IP) port 15450 Video is at (ASTERISK_IP) port 16610 Answering with preferred capability 0x4 (ulaw) phone*CLI> Answering with preferred capability 0x8 (alaw) phone*CLI> Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x200 (speex) phone*CLI> 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) phone*CLI> Answering with non-codec capability 0x1 (telephone-event) phone*CLI> Reliably Transmitting (no NAT) to 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bK15BD Record-Route: Record-Route: From: ;tag=781947608 To: ;tag=as36c4b92b Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 101 INVITE User-Agent: (MYDOMAIN) Phone Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 320 v=0 o=root 19116 19116 IN IP4 (ASTERISK_IP) s=session c=IN IP4 (ASTERISK_IP) t=0 0 m=audio 15450 RTP/AVP 0 8 3 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/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 - - - - --- phone*CLI> -- Executing Playback("SIP/(ACCOUNT2)-5202", "silence/1") in new stack phone*CLI> -- Playing 'silence/1' (language 'en') phone*CLI> <-- SIP read from 216.115.25.198:5061: ACK sip:(ACCOUNT1)@(ASTERISK_IP):5070 SIP/2.0 Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bKF0E From: ;tag=781947608 To: ;tag=as36c4b92b Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 101 ACK Max-Forwards: 13 Content-Length: 0 phone*CLI> --- (10 headers 0 lines)--- phone*CLI> -- Executing BackGround("SIP/(ACCOUNT2)-5202", "(MYDOMAIN)/a2") in new stack phone*CLI> -- Playing '(MYDOMAIN)/a2' (language 'en') phone*CLI> Sep 9 21:57:47 NOTICE[19116]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.18.39.139 phone*CLI> == CDR updated on SIP/(ACCOUNT2)-5202 phone*CLI> -- Executing Goto("SIP/(ACCOUNT2)-5202", "option_1|s|1") in new stack phone*CLI> -- Goto (option_1,s,1) phone*CLI> -- Executing Wait("SIP/(ACCOUNT2)-5202", "1") in new stack phone*CLI> Destroying call '54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com' phone*CLI> Sep 9 21:57:49 NOTICE[19116]: chan_sip.c:4955 sip_reregister: -- Re-registration for (ACCOUNT1)@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/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK529976b9 From: ;tag=as7a6a5ab3 To: Call-ID: 54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com CSeq: 133 REGISTER User-Agent: (MYDOMAIN) Phone Expires: 120 Contact: Event: registration Content-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=z9hG4bK529976b9 From: ;tag=as7a6a5ab3 To: Call-ID: 54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com CSeq: 133 REGISTER Contact: ;expires=20 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Scheduling destruction of call '54265cdd0c03e336258d342a0bed5307@(MYDOMAIN).com' in 32000 ms phone*CLI> Sep 9 21:57:49 NOTICE[19116]: 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> -- Executing Answer("SIP/(ACCOUNT2)-5202", "") in new stack -- Executing BackGround("SIP/(ACCOUNT2)-5202", "enter-ext-of-person") in new stack -- Playing 'enter-ext-of-person' (language 'en') phone*CLI> Destroying call '33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com' phone*CLI> Sep 9 21:57:50 NOTICE[19116]: chan_sip.c:4955 sip_reregister: -- Re-registration for (ACCOUNT2)@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/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2572ab97 From: ;tag=as5150493e To: Call-ID: 33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com CSeq: 133 REGISTER User-Agent: (MYDOMAIN) Phone Expires: 120 Contact: Event: registration Content-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=z9hG4bK2572ab97 From: ;tag=as5150493e To: Call-ID: 33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com CSeq: 133 REGISTER Contact: ;expires=20 Content-Length: 0 phone*CLI> --- (8 headers 0 lines)--- phone*CLI> Scheduling destruction of call '33ce66fe18b7467f43ffab572fe5b865@(MYDOMAIN).com' in 32000 ms phone*CLI> Sep 9 21:57:51 NOTICE[19116]: 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> == CDR updated on SIP/(ACCOUNT2)-5202 phone*CLI> -- Executing Macro("SIP/(ACCOUNT2)-5202", "stde(MYDOMAIN)|302") in new stack phone*CLI> -- Executing Wait("SIP/(ACCOUNT2)-5202", "1") in new stack phone*CLI> -- Executing Playback("SIP/(ACCOUNT2)-5202", "pls-wait-connect-call|skip") in new stack phone*CLI> -- Playing 'pls-wait-connect-call' (language 'en') phone*CLI> -- Executing Ringing("SIP/(ACCOUNT2)-5202", "") in new stack phone*CLI> -- Executing Dial("SIP/(ACCOUNT2)-5202", "SIP/302@(MYDOMAIN).com:5060|20|t") in new stack phone*CLI> -- parse_srv: SRV mapped to host phone.(MYDOMAIN).com, port 5060 phone*CLI> -- Called 302@(MYDOMAIN).com:5060 phone*CLI> -- SIP/(MYDOMAIN).com:5060-7f74 is ringing phone*CLI> -- SIP/(MYDOMAIN).com:5060-7f74 is ringing phone*CLI> -- SIP/(MYDOMAIN).com:5060-7f74 answered SIP/(ACCOUNT2)-5202 -- Attempting native bridge of SIP/(ACCOUNT2)-5202 and SIP/(MYDOMAIN).com:5060-7f74 phone*CLI> == Spawn e(MYDOMAIN)sion (macro-stde(MYDOMAIN), s, 4) exited non-zero on 'SIP/(ACCOUNT2)-5202' in macro 'stde(MYDOMAIN)' == Spawn e(MYDOMAIN)sion (option_1, 302, 1) exited non-zero on 'SIP/(ACCOUNT2)-5202' 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:(CALLERNO)@216.18.39.139:5060 SIP/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK657883aa Route: , From: ;tag=as36c4b92b To: ;tag=781947608 phone*CLI> Contact: Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 102 BYE User-Agent: (MYDOMAIN) Phone Content-Length: 0 --- 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=z9hG4bK657883aa From: ;tag=as36c4b92b To: ;tag=781947608 Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 102 BYE Proxy-Authenticate: Digest realm="216.115.25.198", domain="sip:216.115.25.198", nonce="1341844066", algorithm=MD5 Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (9 headers 0 lines)--- phone*CLI> set_destination: Parsing for address/port to send to phone*CLI> set_destination: set destination to 216.115.25.198, port 5061 phone*CLI> Reliably Transmitting (no NAT) to 216.115.25.198:5061: BYE sip:(CALLERNO)@216.18.39.139:5060 SIP/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK61001d49 Route: , From: ;tag=as36c4b92b To: ;tag=781947608 Contact: Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 103 BYE User-Agent: (MYDOMAIN) Phone Proxy-Authorization: Digest username="(ACCOUNT2)", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="1341844066", response="6dcf6641622a48d8bccde23deda28b0c", opaque="" Date: Sat, 10 Sep 2005 04:57:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 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=z9hG4bK61001d49 From: ;tag=as36c4b92b To: ;tag=781947608 Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 103 BYE Proxy-Authenticate: Digest realm="216.115.25.198", domain="sip:216.115.25.198", nonce="1341844066", algorithm=MD5 Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (9 headers 0 lines)--- phone*CLI> set_destination: Parsing for address/port to send to phone*CLI> set_destination: set destination to 216.115.25.198, port 5061 phone*CLI> Reliably Transmitting (no NAT) to 216.115.25.198:5061: BYE sip:(CALLERNO)@216.18.39.139:5060 SIP/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK2a4b835a Route: , From: ;tag=as36c4b92b To: ;tag=781947608 Contact: Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 104 BYE User-Agent: (MYDOMAIN) Phone Proxy-Authorization: Digest username="(ACCOUNT2)", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="1341844066", response="6dcf6641622a48d8bccde23deda28b0c", opaque="" Date: Sat, 10 Sep 2005 04:57:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 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=z9hG4bK2a4b835a From: ;tag=as36c4b92b To: ;tag=781947608 Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 104 BYE Proxy-Authenticate: Digest realm="216.115.25.198", domain="sip:216.115.25.198", nonce="1341844066", algorithm=MD5 Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (9 headers 0 lines)--- phone*CLI> set_destination: Parsing for address/port to send to phone*CLI> set_destination: set destination to 216.115.25.198, port 5061 phone*CLI> Reliably Transmitting (no NAT) to 216.115.25.198:5061: BYE sip:(CALLERNO)@216.18.39.139:5060 SIP/2.0 Via: SIP/2.0/UDP (ASTERISK_IP):5070;branch=z9hG4bK0807ac3a Route: , From: ;tag=as36c4b92b To: ;tag=781947608 Contact: Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 105 BYE User-Agent: (MYDOMAIN) Phone Proxy-Authorization: Digest username="(ACCOUNT2)", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="1341844066", response="6dcf6641622a48d8bccde23deda28b0c", opaque="" Date: Sat, 10 Sep 2005 04:57:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 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=z9hG4bK0807ac3a From: ;tag=as36c4b92b To: ;tag=781947608 Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 105 BYE Proxy-Authenticate: Digest realm="216.115.25.198", domain="sip:216.115.25.198", nonce="1341844066", algorithm=MD5 Max-Forwards: 15 Content-Length: 0 phone*CLI> --- (9 headers 0 lines)--- phone*CLI> Sep 9 21:57:59 NOTICE[19116]: chan_sip.c:9478 handle_response: Failed to authenticate on BYE to ';tag=781947608' phone*CLI> Destroying call '27558B04-20ED11DA-B03288E9-32645355@216.18.39.139' phone*CLI> <-- SIP read from 216.115.25.198:5061: BYE sip:(ACCOUNT1)@(ASTERISK_IP):5070 SIP/2.0 Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bK13EC From: ;tag=781947608 To: ;tag=as36c4b92b Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 102 BYE Max-Forwards: 13 Content-Length: 0 phone*CLI> --- (10 headers 0 lines)--- phone*CLI> Sending to 216.115.25.198 : 5061 (non-NAT) phone*CLI> Transmitting (no NAT) to 216.115.25.198:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP 216.115.25.198:5061 Via: SIP/2.0/UDP 216.115.20.29:5060 Via: SIP/2.0/UDP 216.18.39.139:5060;branch=z9hG4bK13EC From: ;tag=781947608 To: ;tag=as36c4b92b Call-ID: 27558B04-20ED11DA-B03288E9-32645355@216.18.39.139 CSeq: 102 BYE User-Agent: (MYDOMAIN) Phone Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- phone*CLI> Destroying call '27558B04-20ED11DA-B03288E9-32645355@216.18.39.139' phone*CLI>