[Home]

Summary:ASTERISK-06485: Hanging up during Name Recording Does not Clear Participant
Reporter:Douglas Garstang (dgarstang)Labels:
Date Opened:2006-03-06 10:48:27.000-0600Date Closed:2008-01-15 17:42:11.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents: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