[asterisk-users] One Way Audio

[Tim King]

Thank you I have also tried those settings. The main thing is coming from my
voip provider all I am doing is bridging the calls to two other devices (1
trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with
an IVR and when I call in I can not hear the IVR. However if I call directly
to a SIP client the person answering the SIP phone can hear me but I can not
hear them at all.  Its definately not a NAT issue which is what makes it
even more confusing. When the call is in place a sip show channels shows me
both lefs of the call and they are both using either alaw or ulaw so it
should not be a codec translation issue either.

On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel <satish_lx at hotmail.com> wrote:

> What about your  sip clients? Are they on public network?
>
> Try on sip.conf
>
> Nat=no/yes
>
> conreinvite=yes/no
>
> --
> Sent from my iPhone
>
> On Mar 9, 2011, at 6:11 PM, Tim King <tim at compnetwork.net> wrote:
>
> IPTBALES is off and I have all firewalls disabled. All network elements
> currently involved have public IP's assigned to them. My main asterisk box
> has a public IP. I have multiple trunks to voip peers for inbound and
> outbound calls which are also all public IP's. My two clients are trunked
> via IAX and one is a Trixbox and the other is a digium AA50 which both also
> have public IP's assigned to them.
>
> On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime < <acheraime at gmail.com>
> acheraime at gmail.com> wrote:
>
>> How is your network is organized? Is your server behind a firewal, about
>> iptables ?
>>
>>
>>
>>
>> On Wed, Mar 9, 2011 at 5:40 PM, Tim King < <tim at compnetwork.net>
>> tim at compnetwork.net> wrote:
>>
>>> I am having trouble with no return audio on inbound calls. I have been
>>> working on this for 18 hours and even built a fresh server and moved
>>> everything over and I am getting the same results. I need someone that can
>>> help get this resolved tonight. I can provide access to all machines
>>> involved.
>>>
>>> Please email me at <tim.compnetwork at gmail.com>tim.compnetwork at gmail.comif you can help.
>>>
>>> --
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>>
>>
>>
>> --
>> *Adolphe CHER-AIME
>> Network / VoIP  Engineer
>> CCNA, CCNA VOICE, Global VSAT Forum Certified
>> (509) 3449-4280*
>>
>
> --
> _____________________________________________________________________
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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>               http://www.asterisk.org/hello
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