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

[Peter den Hartog]

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?
>>   
> 
> 
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> 

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)
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