Summary: | ASTERISK-06485: Hanging up during Name Recording Does not Clear Participant | ||
Reporter: | Douglas Garstang (dgarstang) | Labels: | |
Date Opened: | 2006-03-06 10:48:27.000-0600 | Date Closed: | 2008-01-15 17:42:11.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_meetme |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) asterisk-trunk-meetme-recname.patch | |
Description: | Extensions.conf has this: exten => 123,1,Meetme(|dMic|) I dial 123, and enter my conference number. Asterisk asks me to enter my name. At this point I hang up. If I type at the Asterisk console 'meetme list 12345' it shows that I am a participant in the conference evenhough I hung up. If I dial 123 again and this time do not hang up until after I have joined the conference, this does not occur. 'Meetme list' shows 0 participants. The fact that it works the second way and not the first would tend to indicate that it isn't a SIP messaging problem. If Asterisk gets the BYE while I'm in a conference, it should get it when I'm entering a conference. I put this as a MINOR bug, but it's really a huge pain in the ass. ****** ADDITIONAL INFORMATION ****** Console SIP debug output. You can see Asterisk receives the BYE message. It failed to remove the participant from the user. I don't know why that ZAP message is there as I have ztdummy installed and meetme starts up. Destroying call '72add71e5baf07aa6fb92ce7424ba1f0@172.31.140.203' Mar 6 09:46:31 WARNING[23307]: channel.c:2543 ast_request: No channel type registered for 'zap' Mar 6 09:46:31 WARNING[23307]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '12345' -- Recording -- Playing 'vm-rec-name' (language 'en') <-- SIP read from 216.187.140.205:5060: REGISTER sip:ipt.oneeighty.com SIP/2.0 Via: SIP/2.0/UDP 216.187.128.10;branch=z9hG4bKc7dfef14B563665D From: "Test User1" <sip:0220402@ipt.oneeighty.com>;tag=3854088F-ED2E9D3E To: <sip:0220402@ipt.oneeighty.com> CSeq: 2 REGISTER Call-ID: e86208a3-3178c7d1-7df16a38@216.187.128.10 Contact: <sip:0220402@216.187.128.10>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 Authorization: Digest username="9220402", realm="ipt.oneeighty.com", nonce="440c69342343ed73c2561a8ec63156b845cc349d", uri="sip:ipt.oneeighty.com", response="cc1d00478e28167c6d59ddcf8260bdee", algorithm=MD5 Max-Forwards: 69 Expires: 3600 Content-Length: 0 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 216.187.128.10 : 5060 (non-NAT) Transmitting (no NAT) to 216.187.128.10:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 216.187.128.10;branch=z9hG4bKc7dfef14B563665D;received=216.187.140.205 From: "Test User1" <sip:0220402@ipt.oneeighty.com>;tag=3854088F-ED2E9D3E To: <sip:0220402@ipt.oneeighty.com>;tag=as5884e138 Call-ID: e86208a3-3178c7d1-7df16a38@216.187.128.10 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0220402@216.187.140.203> Content-Length: 0 --- Mar 6 09:46:33 NOTICE[23164]: chan_sip.c:10854 handle_request_register: Registration from '<sip:0220402@ipt.oneeighty.com>' failed for '216.187.140.205' - Username/auth name mismatch Scheduling destruction of call 'e86208a3-3178c7d1-7df16a38@216.187.128.10' in 15000 ms <-- SIP read from 216.187.140.205:5060: REGISTER sip:ipt.oneeighty.com SIP/2.0 Via: SIP/2.0/UDP 216.187.128.10;branch=z9hG4bKc7dfef14B563665D From: "Test User1" <sip:0220402@ipt.oneeighty.com>;tag=3854088F-ED2E9D3E To: <sip:0220402@ipt.oneeighty.com> CSeq: 2 REGISTER Call-ID: e86208a3-3178c7d1-7df16a38@216.187.128.10 Contact: <sip:0220402@216.187.128.10>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 Authorization: Digest username="9220402", realm="ipt.oneeighty.com", nonce="440c69342343ed73c2561a8ec63156b845cc349d", uri="sip:ipt.oneeighty.com", response="cc1d00478e28167c6d59ddcf8260bdee", algorithm=MD5 Max-Forwards: 69 Expires: 3600 Content-Length: 0 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 216.187.128.10 : 5060 (non-NAT) Transmitting (no NAT) to 216.187.128.10:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 216.187.128.10;branch=z9hG4bKc7dfef14B563665D;received=216.187.140.205 From: "Test User1" <sip:0220402@ipt.oneeighty.com>;tag=3854088F-ED2E9D3E To: <sip:0220402@ipt.oneeighty.com>;tag=as5884e138 Call-ID: e86208a3-3178c7d1-7df16a38@216.187.128.10 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0220402@216.187.140.203> Content-Length: 0 --- Mar 6 09:46:33 NOTICE[23164]: chan_sip.c:10854 handle_request_register: Registration from '<sip:0220402@ipt.oneeighty.com>' failed for '216.187.140.205' - Username/auth name mismatch Scheduling destruction of call 'e86208a3-3178c7d1-7df16a38@216.187.128.10' in 15000 ms <-- SIP read from 216.187.140.205:5060: BYE sip:123@216.187.140.203 SIP/2.0 Record-Route: <sip:216.187.140.205;ftag=DD527A2D-E1435490;lr=on> Via: SIP/2.0/UDP 216.187.140.205;branch=z9hG4bK00e1.fb63a397.0 Via: SIP/2.0/UDP 216.187.128.68;branch=z9hG4bK33055945208DB97E From: "Douglas Garstang" <sip:2944093@ipt.oneeighty.com>;tag=DD527A2D-E1435490 To: <sip:123@ipt.oneeighty.com;user=phone>;tag=as0d6e2a4e CSeq: 3 BYE Call-ID: c8e3f2c1-8d279edf-2db13e4a@216.187.128.68 Contact: <sip:2944093@216.187.128.68> User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Proxy-Authorization: Digest username="2944093", realm="ipt.oneeighty.com", nonce="440c69291e547e34d15612762eca3ff15abe4701", qop=auth, cnonce="6dYzPYkoW6EoYhA", nc=00000001, uri="sip:123@ipt.oneeighty.com;user=phone", response="6be680dc7c2bf96819ad8e1bc7fcaf53", algorithm=MD5 Max-Forwards: 69 Content-Length: 0 P-hint: rr-enforced --- (14 headers 0 lines)--- Sending to 216.187.140.205 : 5060 (non-NAT) Transmitting (no NAT) to 216.187.140.205:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 216.187.140.205;branch=z9hG4bK00e1.fb63a397.0;received=216.187.140.205 Via: SIP/2.0/UDP 216.187.128.68;branch=z9hG4bK33055945208DB97E Record-Route: <sip:216.187.140.205;ftag=DD527A2D-E1435490;lr=on> From: "Douglas Garstang" <sip:2944093@ipt.oneeighty.com>;tag=DD527A2D-E1435490 To: <sip:123@ipt.oneeighty.com;user=phone>;tag=as0d6e2a4e Call-ID: c8e3f2c1-8d279edf-2db13e4a@216.187.128.68 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:123@216.187.140.203> Content-Length: 0 --- Mar 6 09:46:33 WARNING[23307]: file.c:584 ast_readaudio_callback: Failed to write frame -- Playing 'conf-onlyperson' (language 'en') sip no debug SIP Debugging Disabled | ||
Comments: | By: Douglas Garstang (dgarstang) 2006-03-06 13:14:35.000-0600 Fixed the zap error. It was because I had removed chan_zap.so from modules.conf, not realising I needed zap for meetme() to work. Retested and problem still occurs with this error gone. By: Douglas Garstang (dgarstang) 2006-03-06 13:20:48.000-0600 Also changed the parms to: exten => 123,1,Meetme(|dMic) ... problem persists. By: Joshua C. Colp (jcolp) 2006-03-20 18:08:48.000-0600 Give this patch a whirl, should fix your issue. By: Douglas Garstang (dgarstang) 2006-03-21 09:25:01.000-0600 Awesome. Seems to fix the problem. Thanks. I have no idea... can that patch now be merged into the next release? By: Russell Bryant (russell) 2006-03-21 09:57:06.000-0600 fixed in 1.2 and the trunk in revisions 13851 and 13852, thanks! By: Digium Subversion (svnbot) 2008-01-15 17:42:10.000-0600 Repository: asterisk Revision: 13851 U branches/1.2/apps/app_meetme.c ------------------------------------------------------------------------ r13851 | russell | 2008-01-15 17:42:10 -0600 (Tue, 15 Jan 2008) | 2 lines don't add conference participant if the user hangs up while recording their name (issue ASTERISK-6485) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=13851 By: Digium Subversion (svnbot) 2008-01-15 17:42:11.000-0600 Repository: asterisk Revision: 13852 _U trunk/ U trunk/apps/app_meetme.c ------------------------------------------------------------------------ r13852 | russell | 2008-01-15 17:42:11 -0600 (Tue, 15 Jan 2008) | 10 lines Merged revisions 13851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r13851 | russell | 2006-03-21 10:53:27 -0500 (Tue, 21 Mar 2006) | 2 lines don't add conference participant if the user hangs up while recording their name (issue ASTERISK-6485) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=13852 |