I have seen countless problems resolved by using "notransfer=yes" in IAX.conf stuff like dropped calls, poor quality and even 1 way audio.<br><br><div><span class="gmail_quote">On 11/28/06, <b class="gmail_sendername">
hugolivude</b> <<a href="mailto:hugolivude@gmail.com">hugolivude@gmail.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;">
Asterisk 1.2.7<br>RedHat 9.0<br><br>Hi,<br>I've run into some voice degradation problems with IAX2:<br><br>I frequently have calls come in on a DiD provided by an ITSP. I often<br>have to redirect these calls back out to the PSTN (
i.e. to a cell<br>phone). When this happens, I don't want my server in the media path,<br>I want to hand it off to my ITSP instead and let them handle both ends<br>of the call. I've been very careful to avoid using t or T in my dial
<br>commands.<br><br>I couldn't get this to work with SIP (I'm behind a NAT BTW). No<br>matter how I tried, the media continued to pass through my server, so<br>I switched from SIP to IAX2.<br><br>I've had much better success redirecting calls back to the PSTN using
<br>IAX2. I can see the handshakes in the CLI and once the redirected<br>call had been established, I can phyically disconnect my * server from<br>the Ethernet and the call is unaffected.<br><br>Unfortunately I'm getting complaints about call quality (I use ulaw
<br>the whole way).<br><br>I don't think the problem can possibly be on my server or its<br>configuration given that the call is completly handed off as described<br>above. Surely this must be the ITSP's problem, but perhaps I'm
<br>missing something? Any suggestions are welcome!<br><br>Thanks,<br>H<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>
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