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

[Bogdan-Andrei Iancu]

Hi Peter,

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.
>   
when doing call transfer via REFER, the REFER is propagating to the 
other party and the other party is responsible foe generating the new 
call - but as you have the Asteirsk on the path,it will behave as a end 
point, so * must generate the new call.
> 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)
>   
hmmm...not following here..could you detail a bit?

Regards,
Bogdan