[asterisk-users] Interesting One Way Audio

[Thermal Wetland]

On Tue, Apr 13, 2010 at 7:43 PM, Prince Singh <prince at drishti-soft.com>wrote:

>
>
>    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
>
>


Thank you for the feedback, 4.1 about early media led me to the answer!
Your ideas and voip-info.org searching helped!

my extensions.conf was like this:
answer()
cut()
dial()

I changed it to:
cut()
dial()

Thanks again for your assistance!

-Thermal
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