Summary: | ASTERISK-17077: SIP response don't worked $(HASH(SIP_CAUSE,<slave-channel-name>)} | ||
Reporter: | dogonovmax (dogonovmax) | Labels: | |
Date Opened: | 2010-12-08 06:16:12.000-0600 | Date Closed: | 2011-06-07 14:05:23 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/CallCompletionSupplementaryServices |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | SIP response in asterisk 1.8.0 don't worked for me!!!! ****** ADDITIONAL INFORMATION ****** SIP response in asterisk 1.8.0 don't worked for me!!!! extension.conf [default] exten => _X.,1,Dial(SIP/212.33.19.20/123) exten => _X.,n,Hangup(${HASH(SIP_CAUSE,<channel-name>)}) exten => _X.,n,NoOP(${HASH(SIP_CAUSE,<channel-name>)}) Asterisk logs <--- Transmitting (NAT) to 217.23.69.182:1072 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.23.69.182:1069;branch=z9hG4bK-d8754z-8889c2da84cde9d2-1---d8754z-;received=217.23.69.182;rport=1072 From: <sip:xlite1@91.201.21.181;transport=UDP>;tag=74196722 To: <sip:1234@91.201.21.181;transport=UDP> Call-ID: OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1234@91.201.21.181:5060> Content-Length: 0 <------------> -- Executing [1234@default:1] Dial("SIP/xlite1-00000012", "SIP/212.33.19.20/123") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 212.33.19.20:5060: INVITE sip:123@212.33.19.20 SIP/2.0 Via: SIP/2.0/UDP 91.201.21.181:5060;branch=z9hG4bK689bc2f6 Max-Forwards: 70 From: "xlite1" <sip:xlite1@91.201.21.181>;tag=as07c2ff4a To: <sip:123@212.33.19.20> Contact: <sip:xlite1@91.201.21.181:5060> Call-ID: 5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.0 Date: Wed, 08 Dec 2010 11:26:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 310 v=0 o=root 1351194758 1351194758 IN IP4 91.201.21.181 s=Asterisk PBX 1.8.0 c=IN IP4 91.201.21.181 t=0 0 m=audio 13276 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 - - - - a=ptime:20 a=sendrecv --- -- Called 212.33.19.20/123 <--- SIP read from UDP:212.33.19.20:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.201.21.181:5060;branch=z9hG4bK689bc2f6;received=91.201.21.181 From: "xlite1" <sip:xlite1@91.201.21.181>;tag=as07c2ff4a To: <sip:123@212.33.19.20>;tag=as54a57c33 Call-ID: 5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 212.33.19.20:5060: ACK sip:123@212.33.19.20 SIP/2.0 Via: SIP/2.0/UDP 91.201.21.181:5060;branch=z9hG4bK689bc2f6 Max-Forwards: 70 From: "xlite1" <sip:xlite1@91.201.21.181>;tag=as07c2ff4a To: <sip:123@212.33.19.20>;tag=as54a57c33 Contact: <sip:xlite1@91.201.21.181:5060> Call-ID: 5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.0 Content-Length: 0 --- -- SIP/212.33.19.20-00000013 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [1234@default:2] Hangup("SIP/xlite1-00000012", "") in new stack == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/xlite1-00000012' -- Executing [h@default:1] NoOp("SIP/xlite1-00000012", "") in new stack Scheduling destruction of SIP dialog 'OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 217.23.69.182:1072 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 217.23.69.182:1069;branch=z9hG4bK-d8754z-8889c2da84cde9d2-1---d8754z-;received=217.23.69.182;rport=1072 From: <sip:xlite1@91.201.21.181;transport=UDP>;tag=74196722 To: <sip:1234@91.201.21.181;transport=UDP>;tag=as358b0db1 Call-ID: OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060' Method: INVITE <--- SIP read from UDP:217.23.69.182:1072 ---> ACK sip:1234@91.201.21.181;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 217.23.69.182:1069;branch=z9hG4bK-d8754z-8889c2da84cde9d2-1---d8754z-;rport Max-Forwards: 70 To: <sip:1234@91.201.21.181;transport=UDP>;tag=as358b0db1 From: <sip:xlite1@91.201.21.181;transport=UDP>;tag=74196722 Call-ID: OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU. CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:217.23.69.182:1072 ---> <-------------> Really destroying SIP dialog 'MDEyNmEzMjQ0MmMxZjMzZTYzNjJlNTE1Mjk5OTZlZDQ.' Method: REGISTER Really destroying SIP dialog 'YmQ3MjU1YjQ1OWNkNWU3Mjk2ZmRlMGNlNmU4YWViOGE.' Method: REGISTER Really destroying SIP dialog 'OGNlMjQ4Mjg2NzIwOThmZDFjZTNhMmQ4MWE2NDA4NTI.' Method: REGISTER Really destroying SIP dialog 'OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.' Method: ACK s13*CLI> s13*CLI> s13*CLI> s13*CLI> s13*CLI> s13*CLI> core show version Asterisk 1.8.0 built by root @ s13 on a i686 running Linux on 2010-12-08 09:24:32 UTC <--- SIP read from UDP:217.23.69.182:1072 ---> | ||
Comments: | By: Leif Madsen (lmadsen) 2010-12-16 09:26:36.000-0600 Looks like a support issue to me. Incorrect usage of HASH() or a dialplan that wasn't reloaded. Please use the asterisk-users mailing list or #asterisk IRC channel on the Freenode network. |