[OpenSIPS-Users] Transfer issue

Mon Oct 26 15:28:03 CET 2009

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>
> 
> _______________________________________________
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> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

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