Summary: | ASTERISK-12828: Dial with timeout 0 places a call and immediately cancels it. | ||
Reporter: | Atis Lezdins (atis) | Labels: | |
Date Opened: | 2008-10-06 10:14:55 | Date Closed: | 2008-10-14 18:47:12 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_dial |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Dial should either place a call with no timeout, or don't place it at all. Asterisk 1.6.0 ****** ADDITIONAL INFORMATION ****** [Oct 6 07:26:59] VERBOSE[10657] logger.c: -- Executing [21174@local_dial:88] Dial("SIP/90221-08460530", "SIP/90139,0,gtU(agent_call_answer^21174)") in new stack [Oct 6 07:26:59] DEBUG[10657] chan_sip.c: Initializing initreq for method INVITE - callid 16863c3272e8556b2374575f75129990@192.168.1.80 [Oct 6 07:26:59] VERBOSE[10657] logger.c: Reliably Transmitting (NAT) to 192.168.1.136:5060: INVITE sip:90139@192.168.1.136:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.80:5060;branch=z9hG4bK18eee87f;rport Max-Forwards: 70 From: "TEST Working Place 221" <sip:21169@192.168.1.80>;tag=as4150e010 To: <sip:90139@192.168.1.136:5060> Contact: <sip:21169@192.168.1.80> Call-ID: 16863c3272e8556b2374575f75129990@192.168.1.80 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0 Date: Mon, 06 Oct 2008 14:26:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 469 v=0 o=root 461344733 461344733 IN IP4 192.168.1.80 s=Asterisk PBX 1.6.0 c=IN IP4 192.168.1.80 t=0 0 m=audio 41756 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Oct 6 07:26:59] DEBUG[10657] chan_sip.c: Trying to put 'INVITE sip' onto UDP socket destined for 192.168.1.136:5060 [Oct 6 07:26:59] VERBOSE[10657] logger.c: -- Called 90139 [Oct 6 07:26:59] WARNING[10657] app_dial.c: Invalid timeout specified: '0' [Oct 6 07:26:59] DEBUG[10657] rtp.c: Channel '<unspecified>' has no RTP, not doing anything [Oct 6 07:26:59] DEBUG[10657] channel.c: Hanging up channel 'SIP/90139-082c67d0' [Oct 6 07:26:59] DEBUG[10657] chan_sip.c: Hangup call SIP/90139-082c67d0, SIP callid 16863c3272e8556b2374575f75129990@192.168.1.80 [Oct 6 07:26:59] DEBUG[10657] chan_sip.c: update_call_counter(90139) - decrement call limit counter on hangup [Oct 6 07:26:59] DEBUG[31332] app_queue.c: Device 'SIP/90139' changed to state '1' (Not in use) [Oct 6 07:26:59] DEBUG[10657] chan_sip.c: Acked pending invite 102 [Oct 6 07:26:59] DEBUG[10657] chan_sip.c: Stopping retransmission on '16863c3272e8556b2374575f75129990@192.168.1.80' of Request 102: Match Found [Oct 6 07:26:59] VERBOSE[10657] logger.c: Reliably Transmitting (NAT) to 192.168.1.136:5060: CANCEL sip:90139@192.168.1.136:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.80:5060;branch=z9hG4bK18eee87f;rport Max-Forwards: 70 From: "TEST Working Place 221" <sip:21169@192.168.1.80>;tag=as4150e010 To: <sip:90139@192.168.1.136:5060> Call-ID: 16863c3272e8556b2374575f75129990@192.168.1.80 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.0 Content-Length: 0 | ||
Comments: | By: Leif Madsen (lmadsen) 2008-10-14 11:42:36 Assigning this issue to file as he is probably the best to move this issue forward. I realize he is quite busy, so this issue may take longer to resolve than is typical with him. Please reassign if necessary. Thanks! By: Mark Michelson (mmichelson) 2008-10-14 18:44:36 I am taking the reins on this one. Even though it's totally invalid behavior, I'm not willing to change 1.4 or 1.6.0 even though the change seems to be very logical. Instead, I will add the behavior change to trunk and 1.6.1 so that an invalid timeout will translate to mean no timeout. By: Digium Subversion (svnbot) 2008-10-14 18:47:10 Repository: asterisk Revision: 149279 U trunk/CHANGES U trunk/apps/app_dial.c ------------------------------------------------------------------------ r149279 | mmichelson | 2008-10-14 18:47:10 -0500 (Tue, 14 Oct 2008) | 7 lines When specifying an invalid timeout to Dial, take it to mean that no timeout is desired. (closes issue ASTERISK-12828) Reported by: atis ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=149279 |