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

[James Sizemore]

Make sure you have "canreinvite=yes" in all peers in sip.conf that the 
call goes through.
Also making sure that you don't have "tT" on any of your "Dial" 
statements in extension.conf.

But your real problem is that you have some type of network problem use 
"mii-tool eth0"
at a bash prompt, and make sure you are full duplex on both boxes as 
well as on the switch.
You should be able to have dozens of call chaining through Asterisk 
boxes with out voice
quality problem, even on very modest hardware.



Robert Bedell wrote:

> 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
>