[Asterisk-bsd] Asterisk presence fails

[Richard Neese]

cd /usr/ports/net/asterisk
ee Makefile
change 1.4.10 or 11 to 1.4.13
save
make makesum
make config
do not apply the codec patches.
make 
make deinstall
make install
cd ../asterisk-addons
ee Makefile
change 1.4.2 to 1.4.4
save
make makesum
make install
this will bring you uptodate.



On Wednesday 07 November 2007 17:10:48 Adam Vande More wrote:
> On Nov 6, 2007 3:12 AM, Richard Neese <r.neese at gmail.com> wrote:
> > unless your developing you should never run head. it can be broken at
> > various times.
> >
> > that still does not tell me also if its 1.2 head or 1.4 head.  or 1.6
> > head wich is now in devel.
> >
> > try installing asterisk from ports. and see if you still get the issue.
> >
> > On Tuesday 06 November 2007 15:43:54 you wrote:
> > > On Nov 5, 2007 9:48 AM, Richard Neese <r.neese at gmail.com> wrote:
> > > > your missing info like what ver of asterisk
> > > >
> > > > On Monday 05 November 2007 23:36:07 Adam Vande More wrote:
> > > > > Hello,
> > > > >
> > > > > I'm receiving error's when trying to implement subscriptions on
> > > > > asterisk. Specfically on the failed to extend from x to xx parts. 
> > > > > I can find no examples of dealing with this issue.  The notify
> > > > > packet seems to be sent to the phone, however the content length is
> > > > > 0.  Do anyone have insight?
> > > > >
> > > > > <--- Transmitting (no NAT) to 192.168.0.183:2057 --->
> > > > > SIP/2.0 200 OK^M
> > > > > Via: SIP/2.0/UDP
> > > > > 192.168.0.183:2057;branch=z9hG4bK-nwx9atho5sws;received=
> > > > > 192.168.0.183;rport=2057^M
> > > > > From: <sip:adam at 192.168.0.1>;tag=8bxn28mrm9^M
> > > > > To: <sip:1602 at 192.168.0.1;user=phone>;tag=as5a9e0671^M
> > > > > Call-ID: 3c26720ccd14-vrtja7u2ewou at snom360-000413290953^M
> > > > > CSeq: 2 SUBSCRIBE^M
> > > > > User-Agent: Asterisk PBX SVN-trunk-r84726M^M
> > > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> > > > > NOTIFY^M Supported: replaces^M
> > > > > Expires: 3600^M
> > > > > Contact: <sip:1602 at 192.168.0.1>;expires=3600^M
> > > > > Content-Length: 0^M
> > > > > ^M
> > > > >
> > > > > <------------>
> > > > > [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend from 0
> > > > > to 44 [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend
> > > > > from 0 to 157 [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to
> > > > > extend from 0 to 41 [Nov  5 12:12:52] VERBOSE[1079] logger.c:
> > > > > failed to extend from 0 to 49 [Nov  5 12:12:52] VERBOSE[1079]
> > > > > logger.c: failed to extend from 0 to 48 [Nov  5 12:12:52]
> > > > > VERBOSE[1079] logger.c: Reliably Transmitting (no NAT) to
> > > > > 192.168.0.183:2057:
> > > > > NOTIFY sip:adam at 192.168.0.183:2057;line=gv8x1x75 SIP/2.0^M
> > > > > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5266ef46;rport^M
> > > > > Max-Forwards: 70^M
> > > > > From: <sip:1602 at 192.168.0.1;user=phone>;tag=as5a9e0671^M
> > > > > To: <sip:adam at 192.168.0.1>;tag=8bxn28mrm9^M
> > > > > Contact: <sip:1602 at 192.168.0.1>^M
> > > > > Call-ID: 3c26720ccd14-vrtja7u2ewou at snom360-000413290953^M
> > > > > CSeq: 102 NOTIFY^M
> > > > > User-Agent: Asterisk PBX SVN-trunk-r84726M^M
> > > > > Event: dialog^M
> > > > > Content-Type: application/dialog-info+xml^M
> > > > > Subscription-State: active^M
> > > > > Content-Length: 0^M
> > > > > ^M
> > > > >
> > > > > Thanks
> > >
> > > It was in the packet header....Asterisk PBX SVN-trunk-r84726M.  I'm no
> > > asterisk guru, but I think asterisk is having an issue generating
> > > the xml payload for the packet.  I think that's what these notices are:
> > >
> > > [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend from 0 to 44
> > > [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend from 0 to
> > > 157 [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend from 0
> > > to 41 [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend from 0
> > > to 49 [Nov  5 12:12:52] VERBOSE[1079] logger.c: failed to extend from 0
> > > to 48
> > >
> > > and that's why the content length is 0.
>
> After building the last asterisk from ports, the presence is fully
> functional now.
>
> Thanks,



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