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

[Bogdan-Andrei Iancu]

Peter den Hartog wrote:
>
> 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.
>   
Do you see some NOTIFY requests going around? they are used during 
attended transfer to inform on the new call state.....

Regards,
Bogdan