[Asterisk-Users] repost, SIP - MGCP bridge failing

Ekke Einberg Ekke.Einberg at starman.ee
Fri Jun 27 13:44:07 MST 2003


Of course "canreinvite=no" solves it for outbound calls. But, you don't 
want to have 200 RTP streams going through your * server, do you?
I was hoping that someone knows how to get those endpoints to talk each 
other directly...

e 

John Todd wrote:

>> Hi!
>>
>> Has anyone ever tried to bridge cisco 5300 (talking SIP) and MGCP 
>> endpoint over *?
>> Seems that there is a bug or something. When * reinvites Cisco for 
>> bridge, Cisco replies with different set of SDP parameters and expect 
>> RTP stream on another port.
>>
>> regards,
>> Ekke EInberg
>>
>
> Have you set "canreinvite=no" on both endpoints?  While I have no 
> direct experience with your particular problem, I have seen other RTP 
> mismatches solved through the "kludge" of forcing RTP to always go 
> through Asterisk.
>
> JT
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> .
>





More information about the asterisk-users mailing list