Summary:ASTERISK-13590: Audio is not synchronized and the quality of call is bad
Reporter:watataquinteros (watataquinteros)Labels:
Date Opened:2009-02-16 20:18:03.000-0600Date Closed:2011-06-07 14:02:50
Versions:Frequency of
Description:We have 2 Asterisk (A = and B = 1.4.17): A sends calls to B via SIP and ULAW. We don't have any quality problem with the calls in this case, but when we spy the calls in B, the voice quality is terrible and not synchronized, the only problem is the voice of the agent in B; the original call of A is fine.
The Agents are connected with Polycom 430 to B.
We have tried IAX and we got the same problem, we also tried different codec(g729, GSM)


We are not recording the calls.
The problem occurs without without put the calls on hold
The problem is similar to issue 0012837
Comments:By: Leif Madsen (lmadsen) 2009-02-17 10:31:56.000-0600

The issue you are reporting on Asterisk 1.4.17 is quite old, and a known issue that should be resolved by updating Box B to something newer (such as 1.4.23).

This issue will now be closed. Feel free to re-open if you can reproduce on a newer version of Asterisk. Thanks!

By: watataquinteros (watataquinteros) 2009-02-17 11:28:02.000-0600


Thank you for your help. I forgot to mention that we tried with version and the error persist. I will install the 1.4.23 and I let know you.

By: watataquinteros (watataquinteros) 2009-02-18 10:42:41.000-0600

I installed and the quality the voice improves and sounds syn now, but the supervisor that is spying noted a delay between the voice of our agent and the client, only the supervisor, the client and the operator are fine. Any idea?..
Thank you

By: Joshua C. Colp (jcolp) 2009-02-18 10:58:43.000-0600

How much of a delay are we talking about, and can you elaborate on the delay more? It is normal for there to be a small delay between what the client/operator is saying and what a spy is hearing.

By: watataquinteros (watataquinteros) 2009-02-18 12:30:17.000-0600

Hi file, In 2 minutes the delay is 15-20 seg. Our agents talk 20 minutes average, after that is very difficult to supervise the calls.
The strange thing is that we have this problem when the calls are Asterisk to Asterisk (SIP and ULAW), but when other provider sends calls to us from Freeswitch we don't have this delay. Let me know if I need to do any other special test. Thank you in advance.

By: Joshua C. Colp (jcolp) 2009-05-12 09:06:57

Okay let's see if we can get to the bottom of this. I'm going to need the complete console output with debug set to go to console in logger.conf and "core set debug 1" executed in the console. A general description of the call flow and dialplan would also be useful.

By: Joshua C. Colp (jcolp) 2009-06-02 08:58:04

I'm suspending this issue due to lack of response and I also believe that it has been resolved using the fix from issue 13745. The fix will be in the next release.