Hi,<br>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.<br>
<br>I have my Asterisk server ( running on Debian, <a href="http://192.168.1.101">192.168.1.101</a> ) and Xlite (running on Vista, <a href="http://192.168.1.102">192.168.1.102</a>) on two different machine within the same Lan. My network is ADSL ( home-based ) with a dynamic IP. <br>
<br>When I run the <br>
exten=>222,1,Answer()
<br>
exten=>222,2,Echo()
<br>
exten=>222,3,Hangup()
<br>
<br>
It works as I am getting RTP packet sent and receied and I can hear the echo audio.<br>
<br>
debian*CLI>
<br>
-- Executing [222@my-phones:1] Answer("SIP/2000-b6d06750", "") in new stack
<br>
-- Executing [222@my-phones:2] Echo("SIP/2000-b6d06750", "") in new stack
<br>
Got RTP packet from <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 003468, ts 2904300, len 000160)
<br>
Sent RTP packet to <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 002928, ts 2904296, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 003469, ts 2904460, len 000160)
<br>
Sent RTP packet to <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 002929, ts 2904456, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 003470, ts 2904620, len 000160)
<br>
Sent RTP packet to <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 002930, ts 2904616, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 003471, ts 2904780, len 000160)
<br>
Sent RTP packet to <a href="http://192.168.1.102:42406">192.168.1.102:42406</a> (type 00, seq 002931, ts 2904776, le
<br>
<br>
<br>
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.
<br>
<br>
exten=>333,1,Answer()
<br>
exten=>333,2,Playback(vm-goodbye)
<br>
exten=>333,3,Hangup()
<br>
<br>
<br>
<br>
It does not work and the console output is:
<br>
<br>
-- Executing [333@my-phones:1] Answer("SIP/2000-b6d09708", "") in new stack
<br>
-- Executing [333@my-phones:2] Playback("SIP/2000-b6d09708", "vm-goodbye") in new stack
<br>
Sent RTP packet to <a href="http://192.168.1.102:61588">192.168.1.102:61588</a> (type 00, seq 017315, ts 000160, len 000160)
<br>
-- <SIP/2000-b6d09708> Playing 'vm-goodbye' (language 'en')
<br>
Got RTP packet from <a href="http://192.168.1.102:61588">192.168.1.102:61588</a> (type 00, seq 005474, ts 052000, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:61588">192.168.1.102:61588</a> (type 00, seq 005475, ts 052160, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:61588">192.168.1.102:61588</a> (type 00, seq 005476, ts 052320, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:61588">192.168.1.102:61588</a> (type 00, seq 005477, ts 052480, len 000160)
<br>
Got RTP packet from <a href="http://192.168.1.102:61588">192.168.1.102:61588</a> (type 00, seq 005478, ts 052640, len 000160)<br><br>My sip.conf is like this:<br><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://userid:pass@voipuser.org/userid">userid:pass@voipuser.org/userid</a>
<br>
nat=yes
<br>
externip=<a href="http://58.251.75.233">58.251.75.233</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>
qualify=yes
<br><br><br>Thank you very much for all your kind help.<br><br>Regards,<br>Pete<br>