[asterisk-users] SIP RTP Tunnel

[Ed Greenberg]

Also set canreinvite=no between Asterisk and the provider.

kalle.odenthal at genion.de wrote:
> Hola Sanjay, 
>
> this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1.
>
> Any idea?
>
> Thanx!!
>
> -----Original Message-----
> From: Sanjay Rajdev [mailto:sanjay.rajdev at featherstoneinformatics.com] 
> Sent: Donnerstag, 29. März 2007 18:27
> To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] SIP RTP Tunnel
>
> Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting).
>
> Regards,
> Sanjay Rajdev
>
> ----- Original Message -----
> From: "kalle odenthal" <kalle.odenthal at genion.de>
> To: asterisk-users at lists.digium.com
> Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta
> Subject: [asterisk-users] SIP RTP Tunnel
>
> Hello,
>
> is it possible to rout ALL RTP Data over Asterisk, like
>
> SIP1 <---RTP---> Asterisk <---RTP---> SIP2
>
> I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) 
>
> Thanx, 
>
> Kalle
>
>
>
>
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