| Summary: | ASTERISK-16939: Unattended transfer failure | ||
| Reporter: | Federico No (nofederico) | Labels: | |
| Date Opened: | 2010-11-10 08:42:48.000-0600 | Date Closed: | 2010-11-18 14:03:53.000-0600 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Platform : Debian Lenny Phone: snom firmware 8.4.18 and zoiper OK: phone2 calls phone1, and phone2 press key transfer and dial phone3. OK: phone1 calls phone2, and phone2 press key feature # and dial phone3. Fail: Phone1 calls phone2 and phone2 press key transfer and dial phone3. This works in asterisk 1.8 beta4. ****** ADDITIONAL INFORMATION ****** Reliably Transmitting (no NAT) to 172.17.69.99:5060: OPTIONS sip:42799@172.17.69.99:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK732760f0 Max-Forwards: 70 From: "Asterisk" <sip:Asterisk@172.17.64.65>;tag=as03862ce0 To: <sip:42799@172.17.69.99:5060> Contact: <sip:Asterisk@172.17.64.65:5060> Call-ID: 206646267c3c104225f9fabb2215d9ea@172.17.64.65:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r294467 Date: Wed, 10 Nov 2010 14:12:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.17.69.99:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK732760f0 From: "Asterisk" <sip:Asterisk@172.17.64.65>;tag=as03862ce0 To: <sip:42799@172.17.69.99:5060> Call-ID: 206646267c3c104225f9fabb2215d9ea@172.17.64.65:5060 CSeq: 102 OPTIONS Contact: <sip:42799@172.17.69.99:5060>;reg-id=1 User-Agent: snom300/8.4.18 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '206646267c3c104225f9fabb2215d9ea@172.17.64.65:5060' Method: OPTIONS <--- SIP read from UDP:172.17.64.10:5060 ---> INVITE sip:42795@172.17.64.65;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-c1f2099460498d9f-1---d8754z- Max-Forwards: 70 Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: <sip:42795@172.17.64.65;transport=UDP> From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.7797 Allow-Events: presence, kpml Content-Length: 253 v=0 o=Zoiper_user 0 0 IN IP4 172.17.64.10 s=Zoiper_session c=IN IP4 172.17.64.10 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 12 lines) --- Sending to 172.17.64.10:5060 (no NAT) Using INVITE request as basis request - MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. Found peer '42797' for '42797' from 172.17.64.10:5060 <--- Reliably Transmitting (no NAT) to 172.17.64.10:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-c1f2099460498d9f-1---d8754z-;received=172.17.64.10 From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e To: <sip:42795@172.17.64.65;transport=UDP>;tag=as29385fb8 Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4221c81e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM.' in 6400 ms (Method: INVITE) <--- SIP read from UDP:172.17.64.10:5060 ---> ACK sip:42795@172.17.64.65;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-c1f2099460498d9f-1---d8754z- Max-Forwards: 70 To: <sip:42795@172.17.64.65;transport=UDP>;tag=as29385fb8 From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.17.64.10:5060 ---> INVITE sip:42795@172.17.64.65;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-e461942a59ef7c0c-1---d8754z- Max-Forwards: 70 Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: <sip:42795@172.17.64.65;transport=UDP> From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.7797 Authorization: Digest username="42797",realm="asterisk",nonce="4221c81e",uri="sip:42795@172.17.64.65;transport=UDP",response="5d474fe9ebefc677ff2c9d27011b4169",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 253 v=0 o=Zoiper_user 0 0 IN IP4 172.17.64.10 s=Zoiper_session c=IN IP4 172.17.64.10 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 172.17.64.10:5060 (no NAT) Using INVITE request as basis request - MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. Found peer '42797' for '42797' from 172.17.64.10:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - 0x1002 (gsm|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.17.64.10:8000 Looking for 42795 in internos (domain 172.17.64.65) list_route: hop: <sip:42797@172.17.64.10:5060;transport=UDP> <--- Transmitting (no NAT) to 172.17.64.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-e461942a59ef7c0c-1---d8754z-;received=172.17.64.10 From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e To: <sip:42795@172.17.64.65;transport=UDP> Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42795@172.17.64.65:5060> Content-Length: 0 <------------> -- Executing [42795@internos:1] Dial("SIP/42797-00000004", "sip/42795,60,Tt") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x2 (gsm) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.17.64.199:3072: INVITE sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK7f776926 Max-Forwards: 70 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072> Contact: <sip:42797@172.17.64.65:5060> Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r294467 Date: Wed, 10 Nov 2010 14:12:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer P-Asserted-Identity: "Zoiper" <sip:42797@172.17.64.65> Content-Type: application/sdp Content-Length: 296 v=0 o=root 1486998094 1486998094 IN IP4 172.17.64.65 s=Asterisk PBX SVN-trunk-r294467 c=IN IP4 172.17.64.65 t=0 0 m=audio 14342 RTP/AVP 3 9 101 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 42795 <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK7f776926 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 102 INVITE Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/42795-00000005 is ringing <--- Transmitting (no NAT) to 172.17.64.10:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-e461942a59ef7c0c-1---d8754z-;received=172.17.64.10 From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e To: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42795@172.17.64.65:5060> Content-Length: 0 <------------> <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK7f776926 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 102 INVITE Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/42795-00000005 is ringing <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK7f776926 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 102 INVITE Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/42795-00000005 is ringing <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK7f776926 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 102 INVITE Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 User-Agent: snom821/8.4.18 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 245 v=0 o=root 1684681613 1684681614 IN IP4 172.17.64.199 s=call c=IN IP4 172.17.64.199 t=0 0 m=audio 56264 RTP/AVP 3 9 101 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (13 headers 12 lines) --- Found RTP audio format 3 Found RTP audio format 9 Found RTP audio format 101 Found audio description format gsm for ID 3 Found audio description format g722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x1002 (gsm|g722), peer - audio=0x1002 (gsm|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1002 (gsm|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.17.64.199:56264 list_route: hop: <sip:42795@172.17.64.199:3072> set_destination: Parsing <sip:42795@172.17.64.199:3072> for address/port to send to set_destination: set destination to 172.17.64.199:3072 Transmitting (no NAT) to 172.17.64.199:3072: ACK sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK05ed95ee Max-Forwards: 70 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Contact: <sip:42797@172.17.64.65:5060> Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r294467 Content-Length: 0 --- -- SIP/42795-00000005 answered SIP/42797-00000004 Audio is at 5060 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.17.64.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-e461942a59ef7c0c-1---d8754z-;received=172.17.64.10 From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e To: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42795@172.17.64.65:5060> P-Asserted-Identity: "Snom 821" <sip:42795@172.17.64.65> Content-Type: application/sdp Content-Length: 272 v=0 o=root 1408912260 1408912260 IN IP4 172.17.64.65 s=Asterisk PBX SVN-trunk-r294467 c=IN IP4 172.17.64.65 t=0 0 m=audio 17284 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:172.17.64.10:5060 ---> ACK sip:42795@172.17.64.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.10:5060;branch=z9hG4bK-d8754z-fa55b15f92a2fdc6-1---d8754z- Max-Forwards: 70 Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba From: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 2 ACK User-Agent: Zoiper rev.7797 Authorization: Digest username="42797",realm="asterisk",nonce="4221c81e",uri="sip:42795@172.17.64.65;transport=UDP",response="5d474fe9ebefc677ff2c9d27011b4169",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:172.17.64.199:3072 ---> INVITE sip:42797@172.17.64.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-omjnv5pkhfwl;rport From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 X-Serialnumber: 000413451A49 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.18 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 393 v=0 o=root 1684681613 1684681615 IN IP4 172.17.64.199 s=call c=IN IP4 172.17.64.199 t=0 0 m=audio 56264 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> --- (19 headers 18 lines) --- Sending to 172.17.64.199:3072 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 99 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format g722 for ID 9 Found audio description format g726-32 for ID 99 Found audio description format gsm for ID 3 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x1002 (gsm|g722), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1002 (gsm|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.17.64.199:56264 <--- Transmitting (no NAT) to 172.17.64.199:3072 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-omjnv5pkhfwl;received=172.17.64.199;rport=3072 From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42797@172.17.64.65:5060> Content-Length: 0 <------------> Audio is at 5060 Adding codec 0x2 (gsm) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.17.64.199:3072 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-omjnv5pkhfwl;received=172.17.64.199;rport=3072 From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42797@172.17.64.65:5060> Content-Type: application/sdp Content-Length: 296 v=0 o=root 1486998094 1486998095 IN IP4 172.17.64.65 s=Asterisk PBX SVN-trunk-r294467 c=IN IP4 172.17.64.65 t=0 0 m=audio 14342 RTP/AVP 3 9 101 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/42797-00000004 Retransmitting #1 (no NAT) to 172.17.64.199:3072: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-omjnv5pkhfwl;received=172.17.64.199;rport=3072 From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42797@172.17.64.65:5060> Content-Type: application/sdp Content-Length: 296 v=0 o=root 1486998094 1486998095 IN IP4 172.17.64.65 s=Asterisk PBX SVN-trunk-r294467 c=IN IP4 172.17.64.65 t=0 0 m=audio 14342 RTP/AVP 3 9 101 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly --- <--- SIP read from UDP:172.17.64.199:3072 ---> ACK sip:42797@172.17.64.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-57t8ofl7kwin;rport From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:172.17.64.199:3072 ---> ACK sip:42797@172.17.64.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-57t8ofl7kwin;rport From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:172.17.64.10:5060 ---> <-------------> <--- SIP read from UDP:172.17.64.199:3072 ---> REFER sip:42797@172.17.64.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-vc2ts1p0n9n2;rport From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 2 REFER Max-Forwards: 70 Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Refer-To: sip:42799@172.17.64.65;user=phone Referred-By: sip:42795@172.17.64.65 User-Agent: snom821/8.4.18 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 42799@internos by 42795@172.17.64.65 <--- Transmitting (no NAT) to 172.17.64.199:3072 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-vc2ts1p0n9n2;received=172.17.64.199;rport=3072 From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 2 REFER Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:42797@172.17.64.65:5060> Content-Length: 0 <------------> set_destination: Parsing <sip:42795@172.17.64.199:3072> for address/port to send to set_destination: set destination to 172.17.64.199:3072 Reliably Transmitting (no NAT) to 172.17.64.199:3072: NOTIFY sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK25ad1c14;rport Max-Forwards: 70 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Contact: <sip:42797@172.17.64.65:5060> Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r294467 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 21 SIP/2.0 183 Ringing --- set_destination: Parsing <sip:42795@172.17.64.199:3072> for address/port to send to set_destination: set destination to 172.17.64.199:3072 Reliably Transmitting (no NAT) to 172.17.64.199:3072: NOTIFY sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK78a2d2b3;rport Max-Forwards: 70 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Contact: <sip:42797@172.17.64.65:5060> Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 104 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r294467 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 Ok --- -- Stopped music on hold on SIP/42797-00000004 Scheduling destruction of SIP dialog '1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060' in 6400 ms (Method: REFER) == Spawn extension (internos, 42799, 1) exited non-zero on 'SIP/42797-00000004' Scheduling destruction of SIP dialog 'MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM.' in 6400 ms (Method: ACK) set_destination: Parsing <sip:42797@172.17.64.10:5060;transport=UDP> for address/port to send to set_destination: set destination to 172.17.64.10:5060 Reliably Transmitting (no NAT) to 172.17.64.10:5060: BYE sip:42797@172.17.64.10:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Max-Forwards: 70 From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r294467 Proxy-Authorization: Digest username="42797", realm="asterisk", algorithm=MD5, uri="172.17.64.65", nonce="", response="76b84d3c6d488351c9bd500ded04732f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Retransmitting #1 (no NAT) to 172.17.64.199:3072: NOTIFY sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK25ad1c14;rport Max-Forwards: 70 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Contact: <sip:42797@172.17.64.65:5060> Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r294467 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 21 SIP/2.0 183 Ringing --- Retransmitting #1 (no NAT) to 172.17.64.199:3072: NOTIFY sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK78a2d2b3;rport Max-Forwards: 70 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Contact: <sip:42797@172.17.64.65:5060> Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 104 NOTIFY User-Agent: Asterisk PBX SVN-trunk-r294467 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 16 SIP/2.0 200 Ok --- <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK25ad1c14;rport=5060 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 103 NOTIFY Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Retransmitting #1 (no NAT) to 172.17.64.10:5060: BYE sip:42797@172.17.64.10:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Max-Forwards: 70 From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r294467 Proxy-Authorization: Digest username="42797", realm="asterisk", algorithm=MD5, uri="172.17.64.65", nonce="", response="76b84d3c6d488351c9bd500ded04732f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK78a2d2b3;rport=5060 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 104 NOTIFY Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Retransmitting #2 (no NAT) to 172.17.64.10:5060: BYE sip:42797@172.17.64.10:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Max-Forwards: 70 From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r294467 Proxy-Authorization: Digest username="42797", realm="asterisk", algorithm=MD5, uri="172.17.64.65", nonce="", response="76b84d3c6d488351c9bd500ded04732f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK25ad1c14;rport=5060 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 103 NOTIFY Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK78a2d2b3;rport=5060 From: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce To: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 104 NOTIFY Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:172.17.64.199:3072 ---> BYE sip:42797@172.17.64.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-ywn0v3l1azti;rport From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 User-Agent: snom821/8.4.18 RTP-RxStat: Total_Rx_Pkts=258,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=257,Tx_Pkts=14,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 172.17.64.199:3072 (no NAT) Scheduling destruction of SIP dialog '1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 172.17.64.199:3072 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.199:3072;branch=z9hG4bK-ywn0v3l1azti;received=172.17.64.199;rport=3072 From: <sip:42795@172.17.64.199:3072>;tag=jb6ztv20bg To: "Zoiper" <sip:42797@172.17.64.65>;tag=as19b1c1ce Call-ID: 1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060 CSeq: 3 BYE Server: Asterisk PBX SVN-trunk-r294467 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Retransmitting #3 (no NAT) to 172.17.64.10:5060: BYE sip:42797@172.17.64.10:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Max-Forwards: 70 From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r294467 Proxy-Authorization: Digest username="42797", realm="asterisk", algorithm=MD5, uri="172.17.64.65", nonce="", response="76b84d3c6d488351c9bd500ded04732f" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:172.17.64.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Zoiper rev.7797 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM.' Method: ACK <--- SIP read from UDP:172.17.64.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Zoiper rev.7797 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:172.17.64.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Zoiper rev.7797 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:172.17.64.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK50da2e3b Contact: <sip:42797@172.17.64.10:5060;transport=UDP> To: "Zoiper"<sip:42797@172.17.64.65;transport=UDP>;tag=2b0b1d6e From: <sip:42795@172.17.64.65;transport=UDP>;tag=as1e819fba Call-ID: MzUwNWNhZmVlZWFhNjllMzA0MmNmYjM0NDZiODc1YmM. CSeq: 102 BYE User-Agent: Zoiper rev.7797 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '1b6c4a6252cc784b4efe965f4660b192@172.17.64.65:5060' Method: BYE Reliably Transmitting (no NAT) to 172.17.64.199:3072: OPTIONS sip:42795@172.17.64.199:3072 SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK2660cdde Max-Forwards: 70 From: "Asterisk" <sip:Asterisk@172.17.64.65>;tag=as06cc439d To: <sip:42795@172.17.64.199:3072> Contact: <sip:Asterisk@172.17.64.65:5060> Call-ID: 474c8791138aacd07f29f3e132a595a7@172.17.64.65:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r294467 Date: Wed, 10 Nov 2010 14:13:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.17.64.199:3072 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK2660cdde From: "Asterisk" <sip:Asterisk@172.17.64.65>;tag=as06cc439d To: <sip:42795@172.17.64.199:3072> Call-ID: 474c8791138aacd07f29f3e132a595a7@172.17.64.65:5060 CSeq: 102 OPTIONS Contact: <sip:42795@172.17.64.199:3072>;reg-id=1 User-Agent: snom821/8.4.18 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '474c8791138aacd07f29f3e132a595a7@172.17.64.65:5060' Method: OPTIONS <--- SIP read from UDP:172.17.64.10:5060 ---> <-------------> Reliably Transmitting (no NAT) to 172.17.64.10:5060: OPTIONS sip:42797@172.17.64.10:5060;rinstance=9c663eb591a30090;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK4f3d8bad Max-Forwards: 70 From: "Asterisk" <sip:Asterisk@172.17.64.65>;tag=as022c576d To: <sip:42797@172.17.64.10:5060;rinstance=9c663eb591a30090;transport=UDP> Contact: <sip:Asterisk@172.17.64.65:5060> Call-ID: 2528fa67478daeb54e41a3c87e265a3b@172.17.64.65:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r294467 Date: Wed, 10 Nov 2010 14:13:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.17.64.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.64.65:5060;branch=z9hG4bK4f3d8bad Contact: <sip:172.17.64.10:5060> To: <sip:42797@172.17.64.10:5060;rinstance=9c663eb591a30090;transport=UDP>;tag=c07f5651 From: "Asterisk"<sip:Asterisk@172.17.64.65>;tag=as022c576d Call-ID: 2528fa67478daeb54e41a3c87e265a3b@172.17.64.65:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.7797 Allow-Events: presence, kpml Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '2528fa67478daeb54e41a3c87e265a3b@172.17.64.65:5060' Method: OPTIONS | ||
| Comments: | By: Federico No (nofederico) 2010-11-10 09:26:01.000-0600 Please close this case. Has duplicate 0018192. Thanks. By: Leif Madsen (lmadsen) 2010-11-18 14:03:53.000-0600 Also in the future please attach backtraces as text files, and not inline. | ||