[asterisk-users] Desperately need help with Asterisk setup

[Anselm Martin Hoffmeister]

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