[OpenSIPS-Users] Transfer issue

[Iñaki Baz Castillo]

El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
> Yes, i just noticed that error myself, that's something else didn't had
>  that before today :-), but that's the whole issue, i think it's not
>  sending a new refer, it just creates a new call on line 2, and when i try
>  to press transfer and hang up the call disappears and in my phone screen i
>  see "transfer failed"

This is how attended transfer works:
- A is speaking with B.
- A puts B on hold and sends a *new* INVITE to C (and talks with him).
- A sends a REFER to B with "Refer-To: sip:C at domain;replaces=xxxx".
- B then generates an INVITE to C with "Replaces" header.
- C accepts the call and *replaces* the previous call (established with A) 
since the new INVITE contains a "Replaces" header with previous dialog 
information.


> 
> the situation is like this: 104 is on Asterisk, 105 & 103 are on opensips,
> 104 takes all the outside calls (for now i made it like this, so we are
>  able to transfer the calls announced)
> 
> i call from my mobile, true the sip trunk  to 104. I transfer a call from
> 104 to 105, this works fine. Then i transfer the same call from 105 to 103,
> these last 2 are both opensips extensions..  and that last part, doesn't
> work. the ngrep of a call like this is what you can see in my last post

It's not easy to guess the issue with this information. However, I describe a 
flow that should work:

- Asterisk receives a call and calls to 104 (through OpenSIPS).
- 104 transfers Asterisk to 105, so now Asterisk is speaking with 105.
- 105 transfers Asterisk to 103, so now Asterisk is speaking with 103.

It should work if 105, 103 and 104 have the same configuration for OpenSIPS 
and Asterisk.


 


-- 
Iñaki Baz Castillo <ibc at aliax.net>