[OpenSIPS-Users] Transfer issue

Mon Oct 26 15:44:32 CET 2009

The reason there is no reply because, there is no reply, 
On phone 105, i press Xfer -> 103, 103 rings i pick it up and then i press
the Xfer button again to actualy transfer it. This result in a hangup on
103, and the outside mobile call is still on 105, on line 2.. It's on
hold... i have to manually resume it, or hang it up...

Peter den Hartog wrote:
> 
> I'm sorry, but this is the complete ngrep. I did it like this: ngrep -p -q
> -W byline 5060
> Maby there is no reply, and that's the issue?
> 
> 
> Iñaki Baz Castillo wrote:
>> 
>> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>>> 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
>> 
>> I'm sorry but I cannot help with this trace as it's incomplete. for
>> example 
>> some responses are not shown:
>> 
>> 
>> U 172.16.0.24:5060 -> 172.16.1.14:5060
>> INVITE sip:103 at 172.16.1.14:5060;user=phone SIP/2.0.
>> Via: SIP/2.0/UDP 172.16.0.24;branch=z9hG4bKfa15f43e8B64AABF.
>> From: "105" <sip:105 at 172.16.1.14>;tag=2BB56AE3-ACF37974.
>> To: <sip:103 at 172.16.1.14;user=phone>.
>> CSeq: 1 INVITE.
>> Call-ID: 1cab3410-e688f931-83bcb322 at 172.16.0.24.
>> Contact: <sip:105 at 172.16.0.24>.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
>> NOTIFY, 
>> PRACK, UPDATE, REFER.
>> User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0133.
>> Supported: 100rel,replaces.
>> Allow-Events: talk,hold,conference.
>> Max-Forwards: 70.
>> Content-Type: application/sdp.
>> Content-Length: 247.
>> .
>> v=0.
>> o=- 1256566109 1256566109 IN IP4 172.16.0.24.
>> s=Polycom IP Phone.
>> c=IN IP4 172.16.0.24.
>> t=0 0.
>> a=sendrecv.
>> m=audio 2246 RTP/AVP 8 0 18 101.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:18 G729/8000.
>> a=rtpmap:101 telephone-event/8000.
>> 
>> 
>> U 172.16.0.24:5060 -> 172.16.1.14:5060
>> ACK sip:103 at 172.16.1.14:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 172.16.0.24;branch=z9hG4bKfa15f43e8B64AABF.
>> From: "105" <sip:105 at 172.16.1.14>;tag=2BB56AE3-ACF37974.
>> To: 
>> <sip:103 at 172.16.1.14;user=phone>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ed84.
>> CSeq: 1 ACK.
>> Call-ID: 1cab3410-e688f931-83bcb322 at 172.16.0.24.
>> Contact: <sip:105 at 172.16.0.24>.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
>> NOTIFY, 
>> PRACK, UPDATE, REFER.
>> User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0133.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> 
>> 
>> 
>> Where is the reply there?
>> 
>> 
>>> 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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