Hi,<br><br>My sip.conf has the allow=gsm as shown in the following:<br><br>[general]<br>port = 5060<br>bindaddr = <a href="http://0.0.0.0">0.0.0.0</a><br>context = others<br><br>register =><a href="http://outraspace:password@voipuser.org/outraspace">outraspace:password@voipuser.org/outraspace</a><br>
nat=yes<br>externip=<a href="http://58.251.75.251">58.251.75.251</a><br>localnet=<a href="http://192.168.1.0/255.255.255.0">192.168.1.0/255.255.255.0</a><br>canreinvite=no<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=gsm<br>
qualify=yes<br><br>All the sound files are in /var/lib/asterisk/sounds instead. Is it correct?<br><br>I have tried both Wengo and xlite, but same result. <br><br>I can't figure out what caused the 404 error. Any idea?<br>
<br>Thank you so much for your help.<br><br>Pete<br><br><div class="gmail_quote">On Mon, Mar 17, 2008 at 10:34 PM, Anselm Martin Hoffmeister <<a href="mailto:anselm@hoffmeister-online.de">anselm@hoffmeister-online.de</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay:<br>
> Hi,<br>
><br>
<div class="Ih2E3d">> Here is the SIP debug output for the playback test. Thank you so much<br>
> for your help.<br>
<br>
</div>Hi Pete,<br>
<div class="Ih2E3d"><br>
> <------------><br>
> [Mar 18 05:33:08] -- Executing [333@my-phones:1]<br>
> Answer("SIP/2000-081e0738", "") in new stack<br>
> [Mar 18 05:33:08] Audio is at <a href="http://192.168.1.101" target="_blank">192.168.1.101</a> port 10028<br>
> [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP<br>
> [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP<br>
> [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP<br>
<br>
</div>I do not see "gsm" here. Any reason not to allow that codec? Or did I<br>
miss something? You wrote you enabled it, so it should be here IMO.<br>
<div class="Ih2E3d"><br>
> <--- Transmitting (NAT) to <a href="http://192.168.1.102:5060" target="_blank">192.168.1.102:5060</a> ---><br>
> SIP/2.0 404 Not Found<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:5060" target="_blank">192.168.1.102:5060</a>;branch=z9hG4bK793126083;received=<a href="http://192.168.1.102" target="_blank">192.168.1.102</a>;rport=5060<br>
> From: 2001 <<a href="mailto:sip:2001@192.168.1.101">sip:2001@192.168.1.101</a>>;tag=2612560371<br>
> To: <<a href="mailto:sip:ping@192.168.1.101">sip:ping@192.168.1.101</a>>;tag=as0ca1ddb0<br>
> Call-ID: <a href="mailto:2808830214@192.168.1.102">2808830214@192.168.1.102</a><br>
> CSeq: 20 OPTIONS<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
> Accept: application/sdp<br>
> Content-Length: 0<br>
<br>
</div>"404" does not sound good. Please, look which sound files exist on your<br>
system (e.g. what does<br>
find /usr/share/asterisk -file "vm-goodbye*"<br>
say?)<br>
<br>
Another point: Which client do you use, is it Wengo or is it Xlite? Or<br>
both? In that case: Any differences?<br>
<div><div></div><div class="Wj3C7c"><br>
BR<br>
Anselm<br>
<br>
<br>
<br>
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