Try turning on the Sip debug for the PSTN call as well as RTP debug. Paste the output here.<br><blockquote class="gmail_quote" style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0.8ex; border-left-width: 1px; border-left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; ">
<br><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 255, 255); ">The Dial server is connected to multiple 4-port Redfone devices for<br></span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 255, 255); ">handling PSTN incoming and outgoing calls. Outgoing calls always<br>
</span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 255, 255); ">originate from and incoming calls always terminate at the SIP server.<br></span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 255, 255); ">SIP and Dial servers are connected over IAX2.</span></blockquote>
<div><br></div><div>Explain the above abit as well..couldnt get the clear picture of what it looks like. Seems to me that you guys are using two servers and call-audio gets lost in between the servers OR in between the Dial-Server and redfone device for Queue Calls. </div>
<div> </div><br><div class="gmail_quote">2011/10/29 Raj Mathur (राज माथुर) <span dir="ltr"><<a href="mailto:raju@linux-delhi.org">raju@linux-delhi.org</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote:<br>
> [snip]<br>
<div class="im">> Callers coming in from the PSTN (through the Dial server, over IAX2)<br>
> can also talk normally after an agent has picked up the call.<br>
> However, callers from the PSTN get the announcement and/or MOH<br>
> blanked out after a random period of time, typically 5-10 seconds.<br>
> This often happens in the middle of the queue position or thank-you<br>
> announcement.<br>
><br>
> After the blanking out, the call is still alive, queue functions are<br>
> working, and if an agent picks up the calls s/he can talk normally to<br>
> the caller. However, blanking out of the MOH/announcement makes the<br>
> caller think that the call has been dropped, and they hang up before<br>
> an agent answers.<br>
><br>
> Debug logs show that Asterisk is playing the MOH and announcement<br>
> files continuously even though the caller cannot hear them.<br>
><br>
> Unable to figure out why the blanking happens ONLY on incoming calls<br>
> from the PSTN. Any help appreciated.<br>
<br>
</div>Further simplified the issue to an extension that just does:<br>
<br>
... Answer()<br>
... MusicOnHold(default)<br>
<br>
When called from the PSTN, the musiconhold blanks out after a few<br>
seconds, while it plays fine when the extension is called locally.<br>
<div><div></div><div class="h5"><br>
Regards,<br>
<br>
-- Raj<br>
--<br>
Raj Mathur <a href="mailto:raju@kandalaya.org">raju@kandalaya.org</a> <a href="http://kandalaya.org/" target="_blank">http://kandalaya.org/</a><br>
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