VoIP Mailing List Archives

[dcole at hcit.com.au]

Hello List,

I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up.

For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations.

We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls):

Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK.

This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files.

FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again).

Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0.

I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information.

[general]

disallow=all
allow=g729
mailboxdetail=yes

jitterbuffer=no
;maxjitterbuffer=500
;jittershrinkrate=1
bandwidth=low
tos=lowdelay
trunk=yes
notransfer=yes

#include iax_general_custom.conf
#include iax_registrations_custom.conf
#include iax_registrations.conf
#include iax_custom.conf
#include iax_additional.conf



Any suggestions are very welcome.

Regards,

Daniel
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[pdhales at optusnet.co...]

Does turning off the notransfer help? I would imagine that dropping the
second server out of the equation might be useful, and save some
bandwidth.

PaulH
On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote:

[dcole at hcit.com.au]

Thanks Paul and Lyle for the suggestions.

I would like to keep the phones configuration to one line for now, and see if I can solve the problem rather then just work around it.

I have changed he notransfer option, will see what happens over the next few days.

Thanks again for the suggestions, any further input is very much welcome.
Many Thanks,

Daniel


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul Hales
Sent: Tuesday, 29 January 2008 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Calls - One Way Audio


Does turning off the notransfer help? I would imagine that dropping the second server out of the equation might be useful, and save some bandwidth.

PaulH


On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote: