asterisk users - Mar 2008 - Desperately need help with Asterisk setup

Hi,
I am new to Asterisk and I am having a setup problem that I am trying to
resolved for the last couple days without any success.  I am pretty much
desperated on this issue and I don't know why.  Can someone please kindly
help me to troubleshoot this?  I can't hear any audio from Asterisk when
running Playback or VoiceMail tests.

I have my Asterisk server ( running on Debian, 192.168.1.101 ) and Xlite
(running on Vista, 192.168.1.102) on two different machine  within the same
Lan.  My network is ADSL ( home-based ) with a dynamic IP.

When I run the
exten=>222,1,Answer()
exten=>222,2,Echo()
exten=>222,3,Hangup()

It works as I am getting RTP packet sent and receied and I can hear the echo
audio.

debian*CLI>
-- Executing [222 at my-phones:1] Answer("SIP/2000-b6d06750",
"") in new stack
-- Executing [222 at my-phones:2] Echo("SIP/2000-b6d06750",
"") in new stack
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003468, ts 2904300,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002928, ts 2904296, len
000160)
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003469, ts 2904460,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002929, ts 2904456, len
000160)
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003470, ts 2904620,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002930, ts 2904616, len
000160)
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003471, ts 2904780,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002931, ts 2904776, le


But if I run this, it does not work and I can't hear any of the playback.
from the console, the  packet is not sent to the client.

exten=>333,1,Answer()
exten=>333,2,Playback(vm-goodbye)
exten=>333,3,Hangup()



It does not work and the console output is:

-- Executing [333 at my-phones:1] Answer("SIP/2000-b6d09708",
"") in new stack
-- Executing [333 at my-phones:2] Playback("SIP/2000-b6d09708",
"vm-goodbye")
in new stack
Sent RTP packet to 192.168.1.102:61588 (type 00, seq 017315, ts 000160, len
000160)
-- <SIP/2000-b6d09708> Playing 'vm-goodbye' (language
'en')
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005474, ts 052000, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005475, ts 052160, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005476, ts 052320, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005477, ts 052480, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005478, ts 052640, len
000160)

My sip.conf is like this:


[general]
port = 5060
bindaddr = 0.0.0.0
context = others

register =>userid:pass at voipuser.org/userid
nat=yes
externip=58.251.75.233
localnet=192.168.1.0/255.255.255.0
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
qualify=yes


Thank you very much for all your kind help.

Regards,
Pete
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Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay:
> Hi,
> I am new to Asterisk and I am having a setup problem that I am trying
> to resolved for the last couple days without any success.  I am pretty
> much desperated on this issue and I don't know why.  Can someone
> please kindly help me to troubleshoot this?  I can't hear any audio
> from Asterisk when running Playback or VoiceMail tests.
Dear Pete,

my first idea would be that something with your codecs is borken (TM). I
personally use a setup quite similar to yours, with the one visible
difference that I also allow the "gsm" codec, owing to the fact that
at
least my home-recorded prompts are gsm only. I _guess_ asterisk could or
should handle format conversion from audio files automagically, but for
making sure, please try adding "gsm", at least for now.

You might also want to setup the
[sipclient] stanza in sip.conf such that "nat" is set to
"no", although
I do not see why that should break things. Especially as "Echo" works.

The externip is set to your current external IP, right? (Knowing full
well that some DSL lines get a new IP as often as 6 times a day, or as a
P2P bandwidth countermeasure down to five minute intervals at certain
restrictive providers once your "fair use" volume is used up). Again
this should not be the culprit...

Poking with a stick in the swamps, but perhaps hitting the bug :-P

BR
Anselm