[Asterisk-Users] What happened with the 'reinvitation' on SIP?

[Eric Wieling aka ManxPower]

Megan Willigs wrote:
> Hi everybody
> 
> in new versions of Asterisk the RTP on SIP pass only througt the Asterisk,
> not directly between the endpoints like olders versions.
> 
> What happened whit this feature? (reinvite)
> Can you help me?

The the two legs of the call are using different codecs then reinvites 
won't work.  If you are using t or T option to dial (maybe others), 
Asterisk has to stay in the media stream to listen for the #.  Check 
your codecs.