[OpenSIPS-Users] External transfer fails (from Asterisk)

[Peter den Hartog]

Peter den Hartog wrote:
> 
> 
> 
> Bogdan-Andrei Iancu wrote:
>> 
>> Hi Peter,
>> 
>> Peter den Hartog wrote:
>>> Hello,
>>>
>>> I don't know if i'm on the right mailing list for this issue but maby
>>> i'm
>>> not the only one that had it :-).
>>>   
>> if it is opensips related, you are on the right list :)
>>> I implemented opensips and it works good, the normal calls are going
>>> great,
>>> outside/inside it all works. inside transfer (exten to exten) works to.
>>>
>>> But when an outside caller calls the office, it goes to the asterisk,
>>> and
>>> asterisk forwards it to an opensips extension. exten =
>>> x,Dial,1,(SIP/202 at opensips.org) That works great, the caller gets the
>>> right
>>> person, but when the one being called, transfer that call it gone. 
>>>   
>> This is the first scenario where * is fronting OpenSIPS ...typically is 
>> the other way around :D
>>> I think it's because asterisk is trying to transfer this caller, but the
>>> extension is not there (it's in opensips ofcourse, but not in *) 
>>>   
>> Normally, the call transfer (from the phone) is done via a REFER request 
>> (inside the ongoing dialog) - What I suspect is that , as * is in the 
>> path of all calls with external users, * will intercept the REFER and 
>> try to handle it locally.
>> 
>> Try to get a trace and see if this is what happens = REFER being 
>> consumed by *, instead of passing it to the external party.
>> 
>> Regards,
>> Bogdan
>>> I can connect the asterisk users to the opensips users by connecting the
>>> database, but is this really needed? or is there another issue here? Do
>>> i
>>> miss something?
>>>   
>> 
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
> 
> Hello Bogdan,
> 
> That is correct,
> in Asterisk i see nothing of a new call, or a transfer.. but the phone is
> creating a new call on line 2, in opensips i just see a new ongoing call.
> (the line 2 call) and on the outside phone i hear the asterisk wait/hold
> music.
> 
> Is there any smart solution for this? can i just forward the complete call
> to opensips and let asterisk only forward it, and not create the call? (it
> now just does a dial to the sip member in opensips)
> 


Oke a little update, i can now do blind (cold) transfers from asterisk to
opensips (outside lines) but not hot transfers, then the call gets
disconnected.
-- 
View this message in context: http://n2.nabble.com/External-transfer-fails-from-Asterisk-tp3790325p3793876.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.