[Asterisk-Users] SIP Call transfer with RTP transfer as well?

[Robert Bedell]

I am using SER as a proxy, and using Asterisk as a PBX.  A user calls in to
a 1-800 number.  They listen to the IVR on one Asterisk PBX, and then are
transferred to the call center at the other Asterisk PBX.  Calls are being
brought into the system via SIP.  I need to transfer users from one Asterisk
box to the other.  Functionally this works fine, practically it doesn't as
Asterisk forces the RTP stream to go through the first box into the second.
That kills latency and makes the calls unusable.  Has anyone else had a
similar problem?  I've been looking for a while, and am now fairly
experienced with Asterisk.  Is there a way I don't know of to get Asterisk
to do the SIP call transfer?  Is there a way I can signal back to the SER
proxy not to hang up the call but to transfer it if I can't get Asterisk do
what I want without hacking it?

 

I'm perfectly capable of adding this functionality to Asterisk if necessary,
I just don't want to spend the time if there is already a way to do this.
Maybe I'm doing something stupid and don't realize it.

 

Thanks!

 

Robert

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040503/03263186/attachment.htm