[asterisk-users] IAX Calls - One Way Audio

[Daniel Cole]

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:
> 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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