[asterisk-users] Interesting One Way Audio

[Prince Singh]

   1. Are Asterisk and Mittel in the same physical LAN.. or do they have a
   router between them?
   2. Do a 'rtp debug' at the Asterisk CLI to see where is the RTP data
   being sent to
   3. Probable issues:-
      1. canreinvite is enabled when it should not be
      2. Mitel is sending SDP with an incorrect RTP IP and/or port... You'll
      need to check 'sip debug' to see what RTP port is being sent
   4. From the 1/2 second audio, it seems that it could be due to one of
   these:-
      1. 1/2 second is early media, and is being handled correctly at both
      Mitel and Asterisk. OR,
      2. After 1/2 second, Asterisk and Mitel renogotiate for RTP payload
      type, and switch to a codec that is broken at either or both the locations
      3. After 1/2 second, Asterisk and Mitel renogotiate for RTP IP/port


In case you are unable to debug with the above help, post these:-

   1. IPs of both Mitel and Asterisk
   2. SIP dialog as text (sip debug output should do)
   3. A few lines of RTP debug output

-- 
Regards,
Prince Singh

Drishti-Soft Solutions Pvt Ltd




On Wed, Apr 14, 2010 at 3:56 AM, Thermal Wetland
<thermalwetland at gmail.com>wrote:

> I have an Asterisk box, 1.4.30 with a PRI.
>
> A Mitel 3300 is connected to the Asterisk box via SIP trunking.
>
> When a user calls from the Mitel through the Asterisk box the user can
> speak but can not hear the far end.
>
> But - when I route the Mitel user to echo() it works, send and receive.
> The Mitel user also can record and playback greetings.
>
> One thing I have noticed is that when the Mitel user dials a number that
> autoanswers line 1-800-555-1212 the Mitel user will hear audio for 1/2 a
> second then it is dropped.
>
> I turned of iptables and it acts the same way.
>
> Anyone have any ideas?
>
> --
> -Thermal
>
> --
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