[asterisk-users] Desperately need help with Asterisk setup

[Steve Totaro]

Pete,

You are connecting via a SIP softphone correct?  Where is that in your sip.conf?

On Mon, Mar 17, 2008 at 11:42 AM, Pete Kay <petedao at gmail.com> wrote:
> Hi,
>
> My sip.conf has the allow=gsm as shown in the following:
>
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = others
>
> register =>outraspace:password at voipuser.org/outraspace
>  nat=yes
> externip=58.251.75.251
>
> localnet=192.168.1.0/255.255.255.0
> canreinvite=no
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
>  qualify=yes
>
> All the sound files are in /var/lib/asterisk/sounds instead.  Is it correct?
>
> I have tried both Wengo and xlite, but same result.
>
> I can't figure out what caused the 404 error.  Any idea?
>
>
> Thank you so much for your help.
>
> Pete
>
>
>
> On Mon, Mar 17, 2008 at 10:34 PM, Anselm Martin Hoffmeister
> <anselm at hoffmeister-online.de> wrote:
>
> > Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay:
> > > Hi,
> > >
> >
> > > Here is the SIP debug output for the playback test.  Thank you so much
> > > for your help.
> >
> > Hi Pete,
> >
> >
> > > <------------>
> > > [Mar 18 05:33:08]     -- Executing [333 at my-phones:1]
> > > Answer("SIP/2000-081e0738", "") in new stack
> > > [Mar 18 05:33:08] Audio is at 192.168.1.101 port 10028
> > > [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP
> > > [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP
> > > [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP
> >
> > I do not see "gsm" here. Any reason not to allow that codec? Or did I
> > miss something? You wrote you enabled it, so it should be here IMO.
> >
> >
> > > <--- Transmitting (NAT) to 192.168.1.102:5060 --->
> > > SIP/2.0 404 Not Found
> > > Via: SIP/2.0/UDP
> > >
> 192.168.1.102:5060;branch=z9hG4bK793126083;received=192.168.1.102;rport=5060
> > > From: 2001 <sip:2001 at 192.168.1.101>;tag=2612560371
> > > To: <sip:ping at 192.168.1.101>;tag=as0ca1ddb0
> > > Call-ID: 2808830214 at 192.168.1.102
> > > CSeq: 20 OPTIONS
> > > User-Agent: Asterisk PBX
> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > Supported: replaces
> > > Accept: application/sdp
> > > Content-Length: 0
> >
> > "404" does not sound good. Please, look which sound files exist on your
> > system (e.g. what does
> >        find /usr/share/asterisk -file "vm-goodbye*"
> > say?)
> >
> > Another point: Which client do you use, is it Wengo or is it Xlite? Or
> > both? In that case: Any differences?
> >
> >
> >
> >
> > BR
> > Anselm
> >
> >
> >
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>
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