[asterisk-users] No audio on remote SIP calls

[Ishfaq Malik]

Hi

The only time I've had issues that seem a bit like yours it was down to 
the order of codecs in the handset settings. Make sure they match the 
order dictated on the server.

Ish

Jonathan Moore wrote:
> On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien
> Cramatte<scramatte at zensoluciones.com> wrote:
>   
>> Hi,
>>
>> This sounds udp RTP problem.
>> Might be you have some firewall rules that block this kind of traffic ?
>> As soon I remember, Asterisk by default use random port between 10000
>> and 20000 for rtp traffic (you can adjust this in rtp.conf).
>>     
>
> In theory, there should be no firewalls between my asterisk server
> and the remote phones.  I've opened a ticket with ATT with that
> exact question, as well as a question of rather any NATing is going
> on, though, I doubt this is the case, and this is the first time this type
> of problem has happened in over 4 years.
>
> The idea of RTP being to blame would make sense though.  I can
> still transfer and such, and watching the console, I see when I press
> various keys on the phone, so it seems that the SIP traffic is working
> out fine.  (I do understand that right?  SIP == control RTP == voice
> in a very generic sense?)
>
> I plan to take a packet trace in the morning on the asterisk server and
> see what is going on at that level.  Hints as to what I should be looking
> for?
>
> -jonathan
>
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Ishfaq Malik
Software Developer
PackNet Ltd

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