Is there an easy fix?<br><br>- Dan L.<br><br><div><span class="gmail_quote">On 4/25/06, <b class="gmail_sendername">Philip Edelbrock</b> <<a href="mailto:phil@netroedge.com">phil@netroedge.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>I experienced this today. Doing a 'show channels' in Asterisk showed a<br>Zap line perpetually ringing the sip phone even though the sip phone was<br>reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip
<br>allowed 2-way audio to resume.<br><br><br>Phil<br><br>Frederic Jean wrote:<br>> Hi Geoff,<br>><br>> You might want to try tcdump, specifying the source and destination IP<br>> (to minimize the info)<br>> and see where are the RTP packets going ; you will see if they change
<br>> port or something like that<br>> after a while.<br>><br>> Cheers,<br>> Frederic<br>><br>><br>> ----- Original Message -----<br>> *From:* Geoff Manning <mailto:<a href="mailto:geoffm33@gmail.com">
geoffm33@gmail.com</a>><br>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion<br>> <mailto:<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>> *Sent:* Tuesday, April 25, 2006 17:37
<br>> *Subject:* [Asterisk-Users] One Way Audio....in the middle of a call<br>><br>> We had a user report that they were on a SIP <---> PSTN call for<br>> about 4.5 minutes before the call went to on-way audio. The user
<br>> called the person back and they reported being able to hear my user,<br>> but my user couldn't hear them. The audio condition persisted for<br>> about 15 seconds before the user hung up.<br>>
<br>> Where do I start to troubleshoot one way audio that occurs during a<br>> call?<br>><br>> ------------------------------------------------------------------------<br>><br>> _______________________________________________
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