[asterisk-bugs] [Asterisk 0012443]: Corrupted sound or call recording via IAX and GSM

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12443 
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Reported By:                ztel
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12443
Category:                   Formats/format_gsm
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-14-2008 19:52 CDT
Last Modified:              04-23-2008 14:07 CDT
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Summary:                    Corrupted sound or call recording via IAX and GSM
Description: 
I have a problem with installing Asterisk on Debian / Lenny, exact kernel /
version is:

Linux version 2.6.22-3-686 (Debian 2.6.22-6) (maks at debian.org) (gcc
version 4.1.3 20071019 (prerelease) (Debian 4.1.2-17))
http://bugs.digium.com/view.php?id=1 SMP Mon Nov 12
08:32:57 UTC 2007

STEPS TO REPRODUCE PROBLEM:
1) Download Debian / Lenny which is located at

http://www.us.debian.org/devel/debian-installer/

I use the iso which is located at:

http://cdimage.debian.org/cdimage/lenny_di_beta1/i386/iso-cd/debian-testing-i386-netinst.iso

2) Install on a server with two network cards and make eth0 the WAN and
eth1 the LAN 

3) Set up firewall that can masquerade and do nat... I can provide this
firewall if it helps...

4) install the newest version of asterisk 1.4 on server

5) setup 1 sip device and a voicemail

6) route an extension to this server from a different asterisk server via
IAX to the sip device using either GSM or ULAW.

7) The call will be clear, but the voicemail prompts and recordings will
be garbled or vice versa depending on how you set up the codec in iax.conf

8) change iax.conf to not use gsm and instead use ULAW

9) the call will be garbled but the voicemail prompts will be clear..

there is no way to get them both working..

I have a system set up in this configuration that is currently
experiencing the problem 100% of the time, and this is the 5th different
server i have tried this it is 100% reproducible. 
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Basically the call flow would look like:

INCOMING SIP PROVIDER -> Asterisk TRUNK at datacenter -> application
asterisk server (IAX connect / GSM) -> recoring on local box or forwarding
to sip device

This configuration works great with the Debian Stable (2.6.18) and Debian
Experimantal (2.6.23), but Testing (2.6.22) is broken... 

This is annoying because stable does not work with most SATA controllers
and experimental is too unstable to be a desirable production platform... I
wish I could tell what is different between the versions but I thought it
might be worth reporting to see if any ideas exist.  Google searches do not
result in any significant information to report. 


Please let me know if i can provide any additional information that would
help.

Ztel
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---------------------------------------------------------------------- 
 hesser - 04-23-08 14:07  
---------------------------------------------------------------------- 
Problem confirmed with ubuntu server 8.04 asterisk 1.4.19
Exact same issue. 

Issue History 
Date Modified   Username       Field                    Change               
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04-23-08 14:07  hesser         Note Added: 0085906                          
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