[Asterisk-Users] RTP session between two end users

[Eric Wieling aka ManxPower]

Erdem HAKİ wrote:

> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
> aka ManxPower
> Sent: Monday, June 27, 2005 8:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RTP session between two end users
> 
> Erdem HAKİ wrote:
> 
> 
>>Is it possible that a RTP session between two end users  (so i want to use
>>asterisk as a signaling proxy and bypass RTP sessions)?
>>
>> 
>>
>>I used "canreinvite=yes" but it didn't work. 
>>
>>
>>Description from asterisk conf. File;
>>
>>(canreinvite=yes                ; allow RTP voice traffic to bypass
>>Asterisk)
> 
> 
> 
> It's sip.conf.  reinvites only work if the codec is the same for the 
> two endpoints and Asterisk does NOT have to listen for DTMF (no t or T 
> on the dial line, no meetme, etc.)
> 
> ***************
> We use same codec and don't use meetme etc...  So what else should i do?

How are you determining if RTP audio is going thru Asterisk? 
Remember, SIP signaling will always go thru Asterisk.

Also do a "sip show channels" during a call to confirm that the codecs 
are the same.

-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain