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