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

[Peter den Hartog]

Bogdan-Andrei Iancu wrote:
> 
> 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
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

Nope, no NOTIFY requests. 

Well wat is ment was making asterisk dumb, and just let if forward a
complete call.. so instead of doing a dial to an opensips extention, just
make a full transfer of the call to the opensips server, and then to the
extention.

I'm trying it the other way arround now, as you said earlier that the
opensips recieves all the calls (so is directly connected to the sip trunk)
but i have some strange issue's with that 2, i can't call outside and when i
call inside, the phone rings (i just made a alias) and then i can't pick it
up or anything, the phone doesn't respond!

Any ideas ?

-- 
View this message in context: http://n2.nabble.com/External-transfer-fails-from-Asterisk-tp3790325p3814573.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.