[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

I got rid of that error by putting allowguests=no in asterisk, then i don't
get the error.
i've put a number to a direct opensips extention with
_xxxxxxxxx,Dial,1,(SIP/105 at 172.16.1.14) (.14 beeing my opensips server :))

I called in again, and it ringed @ 105, that works great. Then i do a
anounched transfer from 105 to 103, and again no succes. Again i added the
log:

http://dl.getdropbox.com/u/1382962/grep2.txt


Peter den Hartog wrote:
> 
> Yes, i just noticed that error myself, that's something else didn't had
> that before today :-), but that's the whole issue, i think it's not
> sending a new refer, it just creates a new call on line 2, and when i try
> to press transfer and hang up the call disappears and in my phone screen i
> see "transfer failed" 
> 
> the situation is like this: 104 is on Asterisk, 105 & 103 are on opensips,
> 104 takes all the outside calls (for now i made it like this, so we are
> able to transfer the calls announced) 
> 
> i call from my mobile, true the sip trunk  to 104. I transfer a call from
> 104 to 105, this works fine. Then i transfer the same call from 105 to
> 103, these last 2 are both opensips extensions..  and that last part,
> doesn't work. the ngrep of a call like this is what you can see in my last
> post
> 
> 
> Iñaki Baz Castillo wrote:
>> 
>> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>>> Ok thank you for your clear post, here is the grep of the call i made,
>>> it's
>>>  a outside call to asterisk then a transfer to opensips. (anounched)
>>> that
>>>  one is working, but then i try a transfer from 105 to 103, this is from
>>>  opensips extention to opensips extention. this one fails.
>>> 
>>> http://dl.getdropbox.com/u/1382962/grep.txt
>> 
>> I see no REFER request in that trace.
>> Also what I see is a "484 Addres Incomplete" from Asterisk
>> 
>>> Best regards
>>> 
>>> Iñaki Baz Castillo wrote:
>>> > El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>>> >> Well yes, it does work for the internal calls, but
>>> >> when a call comes in true asterisk to an opensips extention i CAN'T
>>> >>  transfer it :-), i get transfer failed in my screen of my phone, and
>>> >> the call stays on the original called extention. This is only for
>>> >> announced transfers, unannounced works fine.
>>> >>
>>> >> Flavio post stated something about routing your REFER's back to
>>> >> asterisk, so it should work.. but i don't know how to route these
>>> calls
>>> >> back to the
>>> >>  asterisk.
>>> >
>>> > Please, you *already* have the answer. When a phone is speaking with
>>> > Asterisk
>>> > (through OpenSIPS) you must route REFER to Asterisk as *any* other
>>> > in-dialog
>>> > request, this is, the *same* as when a phone is speaking with other
>>> phone
>>> > directly (through OpenSIPS).
>>> >
>>> > If the REFER fails this is because Asterisk is rejecting it !!!
>>> >
>>> > I already suggested you to do a SIP capture (using ngrep) to inspect
>>> > which error replies Asterisk when the REFER arrives to it. Please do
>>> it
>>> > and paste it
>>> > here (I expect a 403 or 404, so it means a wrong configuration in you
>>> > Asterisk, no more).
>>> >
>>> > And please, forget anything about exotic routing of the REFER.
>>> 
>> 
>> 
>> -- 
>> Iñaki Baz Castillo <ibc at aliax.net>
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
> 
> 

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