[OpenSIPS-Users] Transfer issue

Mon Oct 26 15:22:42 CET 2009

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>