[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

I don't have any nat involved here.

so you simply get everything from asterisk to your opensips? and in asterisk
you just have something like:

exten => outsidenumber,1,Dial,(SIP/101 at opensips)?

Because this is how i do it.. and when it reaches my phone this way, i'm
unable to transfer.



Iñaki Baz Castillo wrote:
> 
> El Viernes, 23 de Octubre de 2009, Peter den Hartog escribió:
>> ok, but i want to have my users registered to opensips. And what happend
>> before was this ->
>> call from outside -> asterisk did a dial to 101 in opensips -> 101 ringed
>> i
>> had a call..
>> 
>> transfer this call from 101 to 102 -> i've got a new call on line two
>> from
>> phone 101 to 102, but the call from the outside, was on hold on 101 on
>> line
>> 1..  Blind transfers work perfectly, but not the announced transfers.
>> they
>> just go on hold, and when i press transfer after telling 102 that i have
>> a
>> call from 101, the call just stays on 101 in hold.
>> 
>> How did you fix this? Do you have working announced transfers?
> 
> I didn't fix it as I did never experiment that issue. It *must* be a bug 
> related to you configuration (NAT or whatever).
> 
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

-- 
View this message in context: http://n2.nabble.com/Transfer-issue-tp3877950p3878794.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.