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Summary:ASTERISK-05875: 3-way conferences with SNOM phones and a Zaptel channel do not work
Reporter:Jan-Peter Koopmann (jkoopmann)Labels:
Date Opened:2005-12-20 07:11:31.000-0600Date Closed:2006-01-31 00:16:39.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
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Description:We are using SNOM (www.snom.com) phones with Asterisk 1.2.0. Generally the conferencing functionality of the phone works nicely (press hold, call the second person then press the conference button). If however one of the participants is using zaptel and the other two are using SIP, the remote SIP user cannot be understood after a few seconds (hacked sound etc.).

Example:

      SNOM1 (setting up the conference)
   *zap    *SIP
PSTN1        SNOM2

SNOM1 and PSTN1 can talk without problems but no one is able to understand SNOM2 whereas SNOM2 is able to understand the others correctly. All SIP phones are defined with canreinvite=no. This also happens if SNOM2 is substituted with other SIP devices.

I suspect this is some sort of RTP issue but I have no clue on how to debug this. Therefore if anyone could possibly give me a clue on what you need to debug stuff, please let me know!
Comments:By: Tilghman Lesher (tilghman) 2005-12-20 07:19:57.000-0600

If you establish the conference using app_meetme, do you have problems?  I suspect the mixing is being done on your SNOM phone itself.  Also, the output of 'show channels' while this problem is occurring would be helpful.
By: Olle Johansson (oej) 2005-12-20 07:27:09.000-0600

Pressing the conference button means that your phone is doing the conference, not Asterisk
By: Mark Spencer (markster) 2005-12-20 07:47:41.000-0600

Not an Asterisk issue, please contact SNOM.
By: Jan-Peter Koopmann (jkoopmann) 2005-12-20 07:51:37.000-0600

I was just about to enlarge on this topic when it already was closed... :-)

I agree that the SNOM is doing the conferencing. app_meetme works fine, show channels confirms two channels to the SNOM. I will open a trouble ticket with SNOM immediatly.

However the SNOM has no problem doing this with other VoIP servers. Up to this point the problem is only know to happen in the described scenario that is with Asterisk envolved. This might indicate something being wrong with interop here after all, does it not?
By: Olle Johansson (oej) 2005-12-20 07:54:23.000-0600

Are normal calls between the SNOM and Asterisk to a Zaptel line without quality problems? Does this *only* happen when you use a local conference?
By: Jan-Peter Koopmann (jkoopmann) 2005-12-20 07:56:44.000-0600

I just rechecked and was able to reproduce the problem with three SIP participants as well. Therefore this seems to be a SNOM/Asterisk/SIP problem only and we can forget the zaptel part.

To answer your question: Yes other calls are without quality problems (set aside the usual temporary echo problems once in a while).
By: Olle Johansson (oej) 2005-12-21 01:06:49.000-0600

This must be a phone issue. If normal calls work all right, but conference calls does not work properly, I guess that the difference between the calls is on the phone side, not the Asterisk side.

Wonder if SNOM enables silence suppression or something else on those calls.
By: Olle Johansson (oej) 2005-12-21 01:07:34.000-0600

We need a packet trace for a normal call and a conference call. Turn debug to level 4, verbosity to 4 and enable SIP debugging. Capture both calls from start to end and upload that file here.
By: Serge Vecher (serge-v) 2005-12-21 09:29:50.000-0600

oej: there might be more to this than the phone issue. I've observed one of my Asterisk systems two weeks ago going into a deadlock after one of the users used a Conference button on a Cisco phone. I wasn't able to capture either a sip trace or a backtrace, so I didn't file a bug report. So far, the problem has not resurfaced though...
By: Jan-Peter Koopmann (jkoopmann) 2005-12-21 10:20:19.000-0600

oej: I will be able to produce the debug data on 12/23/05 and will upload it ASAP. Thanks. Just upgraded to 1.2.1. Let's see what this brings. :-)
By: Jan-Peter Koopmann (jkoopmann) 2005-12-23 10:26:52.000-0600

Sorry I was not able to produce the debugs but bug 6048 took all my time today. I am on vacation until January 2nd and will try to catch up with this case ASAP.
By: Olle Johansson (oej) 2006-01-03 05:00:16.000-0600

Welcome back. Any progress?
By: Jan-Peter Koopmann (jkoopmann) 2006-01-04 09:47:26.000-0600

Hi. I am back and now that issue 6048 is solved as well I can finally continue on this one here.

oej: Please allow a dumb question. When you say "capture the calls" do you mean just the debug/console output or the actual IP traffic associated with the call? If so: How would you like it? Ethereal packets?

Kind regards,
 JP
By: Jan-Peter Koopmann (jkoopmann) 2006-01-11 04:27:03.000-0600

Hi oej: Had any chance to look at my question yet?

Kind regards,
 JP
By: Olle Johansson (oej) 2006-01-11 09:17:17.000-0600

I need to see what's going on in Asterisk. Turn on sip history, turn on dumphistory=yes in sip.conf and set debug and verbosity to 4, turn on sip debug and capture all of it in a big messy file.

Thanks!
By: Olle Johansson (oej) 2006-01-11 09:17:17.000-0600

I need to see what's going on in Asterisk. Turn on sip history, turn on dumphistory=yes in sip.conf and set debug and verbosity to 4, turn on sip debug and capture all of it in a big messy file.

Thanks!
By: Olle Johansson (oej) 2006-01-11 09:17:28.000-0600

I need to see what's going on in Asterisk. Turn on sip history, turn on dumphistory=yes in sip.conf and set debug and verbosity to 4, turn on sip debug and capture all of it in a big messy file.

Thanks!
By: Olle Johansson (oej) 2006-01-30 12:19:53.000-0600

,,,watiing for more information. If I can't get it soon, we will have to close this issue. /Olle
By: Jan-Peter Koopmann (jkoopmann) 2006-01-31 00:10:15.000-0600

I understand. We tried to reproduce the bug yesterday but failed to do so for an yet unknown reason. We are going to try again this afternoon. Maybe the bug "went" away during the upgrades from 1.2.0 to 1.2.2 and the firmware upgrades of the snom (now 5.2).

If we cannot reproduce it this afternoon I would suggest closing this for now. If it happens again in the future I can always reopen the issue and then attach the sip-history.

Ok?
By: Olle Johansson (oej) 2006-01-31 00:16:38.000-0600

Yes, please re-open if it happens again. Thanks. /Olle