Hi,<br>I switched to Wengo and solved the one "beat"problem. However, I am still not able to listen to the recorded .wav sound. Can anyone please point me to the right direction? How to listen to the .wav sound?<br>
<br>Thanks,<br>Pete<br><br><div class="gmail_quote">On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas <<a href="mailto:crt.rojas@gmail.com">crt.rojas@gmail.com</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;">
Hello,<br><br>Do your verify, the codecs, of both clients, in your sip.conf?<br><br>What codec do you use?<br><br>Best Regards<br><br><div class="gmail_quote"><div><div></div><div class="Wj3C7c">On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay <<a href="mailto:petedao@gmail.com" target="_blank">petedao@gmail.com</a>> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">Hi,<br>I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can.<br>
<br>Problem 1:<br><br>I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a "beat" sounds, and then nothing else. In the console, I can see:<br>
*CLI> -- Executing [111@my-phones:1] Answer("SIP/2001-081dd6e0", "") in new stack<br> -- Executing [111@my-phones:2] VoiceMail("SIP/2001-081dd6e0", "2000") in new stack<br>
Sent RTP packet to <a href="http://58.251.75.228:9956" target="_blank">58.251.75.228:9956</a> (type 00, seq 037718, ts 000160, len 000160)<br> -- <SIP/2001-081dd6e0> Playing 'vm-intro' (language 'en')<br>
<span style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">Sent RTP packet to <a href="http://58.251.75.228:9956" target="_blank">58.251.75.228:9956</a> (type 00, seq 037719, ts 000320, len 000160)</span><br style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">
<span style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">Sent RTP packet to <a href="http://58.251.75.228:9956" target="_blank">58.251.75.228:9956</a> (type 00, seq 037720, ts 000480, len 000160)</span><br style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">
<span style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">Sent RTP packet to <a href="http://58.251.75.228:9956" target="_blank">58.251.75.228:9956</a> (type 00, seq 037721, ts 000640, len 000160)</span><br style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">
<span style="background-color: rgb(255, 255, 255); color: rgb(255, 0, 0);">Got RTP packet from <a href="http://192.168.1.102:8000" target="_blank">192.168.1.102:8000</a> (type 00, seq 062222, ts 1373137124, len 000160)</span><br style="background-color: rgb(204, 0, 0); color: rgb(255, 255, 255);">
Sent RTP packet to <a href="http://192.168.1.102:8000" target="_blank">192.168.1.102:8000</a> (type 00, seq 037722, ts 000800, len 000160)<br>Sent RTP packet to <a href="http://192.168.1.102:8000" target="_blank">192.168.1.102:8000</a> (type 00, seq 037723, ts 000960, len 000160)<br>
<br>Is it the prolem? First it sends to the public address of the the router, then it sends to the virtual IP. Is this the problem that causing my to hear just one "beat" sound and then no audio? <br><br>Problem 2:<br>
<br>The problem is isolated from Problem 1, cuz I run the SIP client on the same machine as the server, so there should not be network problem. I recorded some voice mails and they are stored as .wav files ok. When I tried to hear back the message, It does not work. Is there any configuration that I have to go through to have Asterisk to play .wav file? <br>
<br>Thank you very much in advance for all your kind help.<br><font color="#888888"><br>Pete<br><br>
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