bernard*CLI> sip debug SIP Debugging enabled bernard*CLI> <-- SIP read from 134.96.249.200:2051: INVITE sip:2@134.96.249.48;user=phone SIP/2.0 Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-w11uxfomfwev;rport From: ;tag=ihd8obxhex To: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 1 INVITE Max-Forwards: 70 Contact: P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600 Content-Type: application/sdp Content-Length: 372 v=0 o=root 1182873160 1182873160 IN IP4 134.96.249.200 s=call c=IN IP4 134.96.249.200 t=0 0 m=audio 49294 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (17 headers 17 lines)--- Using INVITE request as basis request - 3c2670140753-8sgb2r5kzda1@snom360 Sending to 134.96.249.200 : 2051 (NAT) Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:7155 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 134.96.249.200:2051: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-w11uxfomfwev;rport;received=134.96.249.200 From: ;tag=ihd8obxhex To: ;tag=as22b010de Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5638ae93" Content-Length: 0 --- Scheduling destruction of call '3c2670140753-8sgb2r5kzda1@snom360' in 15000 ms Found user '1' bernard*CLI> <-- SIP read from 134.96.249.200:2051: ACK sip:2@134.96.249.48;user=phone SIP/2.0 Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-w11uxfomfwev;rport From: ;tag=ihd8obxhex To: ;tag=as22b010de Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 1 ACK Max-Forwards: 70 Contact: Content-Length: 0 --- (9 headers 0 lines)--- Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c2670140753-8sgb2r5kzda1@snom360' of Response 1: Match Found <-- SIP read from 134.96.249.200:2051: INVITE sip:2@134.96.249.48;user=phone SIP/2.0 Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-rg7w16uvr444;rport From: ;tag=ihd8obxhex To: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 2 INVITE Max-Forwards: 70 Contact: P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600 Proxy-Authorization: Digest username="1",realm="asterisk",nonce="5638ae93",uri="sip:2@134.96.249.48;user=phone",response="db44d113950d42fd40f21bd99dc0c3bd",algorithm=md5 Content-Type: application/sdp Content-Length: 372 v=0 o=root 1182873160 1182873160 IN IP4 134.96.249.200 s=call c=IN IP4 134.96.249.200 t=0 0 m=audio 49294 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (18 headers 17 lines)--- Using INVITE request as basis request - 3c2670140753-8sgb2r5kzda1@snom360 Sending to 134.96.249.200 : 2051 (NAT) Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:7155 check_user_full: Setting NAT on RTP to 0 Found user '1' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 134.96.249.200:49294 Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 134.96.249.200:49294 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device 1 Looking for 2 in default (domain 134.96.249.48) Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:6137 build_route: build_route: Contact hop: list_route: hop: Transmitting (no NAT) to 134.96.249.200:2051: SIP/2.0 100 Trying Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-rg7w16uvr444;rport;received=134.96.249.200 From: ;tag=ihd8obxhex To: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/1-abcb", "SIP/2") in new stack Jun 7 17:02:45 DEBUG[14832]: chan_sip.c:1874 create_addr_from_peer: Setting NAT on RTP to 0 Jun 7 17:02:45 DEBUG[14832]: chan_sip.c:2068 sip_call: Outgoing Call for 2 We're at 134.96.249.48 port 12452 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 134.96.249.247:2051: INVITE sip:2@134.96.249.247:2051;line=oqro9kg2 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK47d75b34;rport From: "1" ;tag=as7fd9a969 To: Contact: Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Jun 2006 15:02:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 265 v=0 o=root 14716 14716 IN IP4 134.96.249.48 s=session c=IN IP4 134.96.249.48 t=0 0 m=audio 12452 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 2 bernard*CLI> <-- SIP read from 134.96.249.247:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK47d75b34;rport=5060 From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 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)--- Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '01c5dae514cef1c41d2488642570d89d@134.96.249.48' Request 102: Found -- SIP/2-bf5c is ringing Transmitting (no NAT) to 134.96.249.200:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-rg7w16uvr444;rport;received=134.96.249.200 From: ;tag=ihd8obxhex To: ;tag=as0490d327 Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- bernard*CLI> <-- SIP read from 134.96.249.247:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK47d75b34;rport=5060 From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 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)--- Jun 7 17:02:45 DEBUG[14765]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '01c5dae514cef1c41d2488642570d89d@134.96.249.48' Request 102: Found -- SIP/2-bf5c is ringing bernard*CLI> <-- SIP read from 134.96.249.247:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK47d75b34;rport=5060 From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 102 INVITE Contact: User-Agent: snom360/4.4 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 212 v=0 o=root 1719976532 1719976533 IN IP4 134.96.249.247 s=call c=IN IP4 134.96.249.247 t=0 0 m=audio 57912 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 10 lines)--- Jun 7 17:02:46 DEBUG[14765]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jun 7 17:02:46 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '01c5dae514cef1c41d2488642570d89d@134.96.249.48' of Request 102: Match Found Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 134.96.249.247:57912 Jun 7 17:02:46 DEBUG[14765]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 134.96.249.247:57912 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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) Jun 7 17:02:46 DEBUG[14765]: chan_sip.c:6137 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.247, port 2051 Transmitting (no NAT) to 134.96.249.247:2051: ACK sip:2@134.96.249.247:2051;line=oqro9kg2 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK4f64a4b7;rport From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Contact: Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/2-bf5c answered SIP/1-abcb We're at 134.96.249.48 port 13034 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 134.96.249.200:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-rg7w16uvr444;rport;received=134.96.249.200 From: ;tag=ihd8obxhex To: ;tag=as0490d327 Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 14716 14716 IN IP4 134.96.249.48 s=session c=IN IP4 134.96.249.48 t=0 0 m=audio 13034 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1-abcb and SIP/2-bf5c set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.247, port 2051 We're at 134.96.249.48 port 12452 Adding codec 0x4 (ulaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 16 lines Reliably Transmitting (no NAT) to 134.96.249.247:2051: INVITE sip:2@134.96.249.247:2051;line=oqro9kg2 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK25853fa2;rport From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Contact: Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 369 v=0 o=root 14716 14717 IN IP4 134.96.249.200 s=session c=IN IP4 134.96.249.200 t=0 0 m=audio 49294 RTP/AVP 0 4 3 8 111 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- bernard*CLI> <-- SIP read from 134.96.249.200:2051: ACK sip:2@134.96.249.48 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.200:2051;branch=z9hG4bK-0z4e69a7uoo9;rport From: ;tag=ihd8obxhex To: ;tag=as0490d327 Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 2 ACK Max-Forwards: 70 Contact: Content-Length: 0 --- (9 headers 0 lines)--- Jun 7 17:02:46 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c2670140753-8sgb2r5kzda1@snom360' of Response 2: Match Found Jun 7 17:02:46 DEBUG[14765]: chan_sip.c:9567 check_pendings: Sending pending reinvite on '3c2670140753-8sgb2r5kzda1@snom360' set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.200, port 2051 We're at 134.96.249.48 port 13034 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 134.96.249.200:2051: INVITE sip:1@134.96.249.200:2051 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK296935dc;rport From: ;tag=as0490d327 To: ;tag=ihd8obxhex Contact: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 220 v=0 o=root 14716 14717 IN IP4 134.96.249.247 s=session c=IN IP4 134.96.249.247 t=0 0 m=audio 57912 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- bernard*CLI> <-- SIP read from 134.96.249.247:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK25853fa2;rport=5060 From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 103 INVITE Contact: User-Agent: snom360/4.4 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 212 v=0 o=root 1719976532 1719976534 IN IP4 134.96.249.247 s=call c=IN IP4 134.96.249.247 t=0 0 m=audio 57912 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 10 lines)--- Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:1379 __sip_ack: Acked pending invite 103 Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '01c5dae514cef1c41d2488642570d89d@134.96.249.48' of Request 103: Match Found Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 134.96.249.247:57912 Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 134.96.249.247:57912 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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) Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:6080 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.247, port 2051 Transmitting (no NAT) to 134.96.249.247:2051: ACK sip:2@134.96.249.247:2051;line=oqro9kg2 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK4deace6b;rport From: "1" ;tag=as7fd9a969 To: ;tag=m8ba54qscn Contact: Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- bernard*CLI> <-- SIP read from 134.96.249.200:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK296935dc;rport=5060 From: ;tag=as0490d327 To: ;tag=ihd8obxhex Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom360/4.4 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer upported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 1182873160 1182873161 IN IP4 134.96.249.200 s=call c=IN IP4 134.96.249.200 t=0 0 m=audio 49294 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c2670140753-8sgb2r5kzda1@snom360' of Request 102: Match Found Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 134.96.249.200:49294 Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 134.96.249.200:49294 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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) Jun 7 17:02:47 DEBUG[14765]: chan_sip.c:6137 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.200, port 2051 Transmitting (no NAT) to 134.96.249.200:2051: ACK sip:1@134.96.249.200:2051;line=yf0er7cf SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK4e817db3;rport From: ;tag=as0490d327 To: ;tag=ihd8obxhex Contact: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 7 17:02:50 DEBUG[14765]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call '3c2670099eb1-yfxs300m2rzm@snom360' Destroying call '3c2670099eb1-yfxs300m2rzm@snom360' bernard*CLI> <-- SIP read from 134.96.249.247:2051: BYE sip:1@134.96.249.48 SIP/2.0 Via: SIP/2.0/UDP 134.96.249.247:2051;branch=z9hG4bK-m314wxe3me8z;rport From: ;tag=m8ba54qscn To: "1" ;tag=as7fd9a969 Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 1 BYE Max-Forwards: 70 Contact: User-Agent: snom360/4.4 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 134.96.249.247 : 2051 (NAT) Transmitting (NAT) to 134.96.249.247:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 134.96.249.247:2051;branch=z9hG4bK-m314wxe3me8z;received=134.96.249.247;rport=2051 From: ;tag=m8ba54qscn To: "1" ;tag=as7fd9a969 Call-ID: 01c5dae514cef1c41d2488642570d89d@134.96.249.48 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.200, port 2051 We're at 134.96.249.48 port 13034 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 134.96.249.200:2051: INVITE sip:1@134.96.249.200:2051;line=yf0er7cf SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK2cd4c14e;rport From: ;tag=as0490d327 To: ;tag=ihd8obxhex Contact: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 218 v=0 o=root 14716 14718 IN IP4 134.96.249.48 s=session c=IN IP4 134.96.249.48 t=0 0 m=audio 13034 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 7 17:02:51 DEBUG[14832]: channel.c:3480 ast_channel_bridge: Returning from native bridge, channels: SIP/1-abcb, SIP/2-bf5c Jun 7 17:02:51 DEBUG[14832]: chan_sip.c:2426 sip_hangup: update_call_counter(2) - decrement call limit counter Jun 7 17:02:51 DEBUG[14832]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=ANSWER. == Spawn extension (default, 2, 1) exited non-zero on 'SIP/1-abcb' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'default' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/1-abcb' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/2-bf5c' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Dial' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/2' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-06-07 17:02:45' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-06-07 17:02:46' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-06-07 17:02:51' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '6' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '5' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1149692565.0' Jun 7 17:02:51 DEBUG[14832]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jun 7 17:02:51 DEBUG[14832]: chan_sip.c:2426 sip_hangup: update_call_counter(1) - decrement call limit counter bernard*CLI> <-- SIP read from 134.96.249.200:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK2cd4c14e;rport=5060 From: ;tag=as0490d327 To: ;tag=ihd8obxhex Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 103 INVITE Contact: Session-Expires: 3600 User-Agent: snom360/4.4 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer upported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 224 v=0 o=root 1182873160 1182873162 IN IP4 134.96.249.200 s=call c=IN IP4 134.96.249.200 t=0 0 m=audio 49294 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Jun 7 17:02:51 DEBUG[14765]: chan_sip.c:1379 __sip_ack: Acked pending invite 103 Jun 7 17:02:51 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c2670140753-8sgb2r5kzda1@snom360' of Request 103: Match Found Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 134.96.249.200:49294 Jun 7 17:02:51 DEBUG[14765]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 134.96.249.200:49294 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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) Jun 7 17:02:51 DEBUG[14765]: chan_sip.c:6080 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.200, port 2051 Transmitting (no NAT) to 134.96.249.200:2051: ACK sip:1@134.96.249.200:2051;line=yf0er7cf SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK78973781;rport From: ;tag=as0490d327 To: ;tag=ihd8obxhex Contact: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 134.96.249.200, port 2051 Reliably Transmitting (no NAT) to 134.96.249.200:2051: CANCEL sip:1@134.96.249.200:2051;line=yf0er7cf SIP/2.0 Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK78973781;rport From: ;tag=as0490d327 To: ;tag=ihd8obxhex Contact: Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '3c2670140753-8sgb2r5kzda1@snom360' in 32000 ms Destroying call '01c5dae514cef1c41d2488642570d89d@134.96.249.48' Jun 7 17:02:51 DEBUG[14765]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call '3c26700b2981-mvbt3j767te2@snom360' Destroying call '3c26700b2981-mvbt3j767te2@snom360' bernard*CLI> <-- SIP read from 134.96.249.200:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 134.96.249.48:5060;branch=z9hG4bK78973781;rport=5060 From: ;tag=as0490d327 To: ;tag=ihd8obxhex Call-ID: 3c2670140753-8sgb2r5kzda1@snom360 CSeq: 103 CANCEL Content-Length: 0 --- (7 headers 0 lines)--- Jun 7 17:02:51 DEBUG[14765]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c2670140753-8sgb2r5kzda1@snom360' of Request 103: Match Found Destroying call '3c2670140753-8sgb2r5kzda1@snom360'