Norbert Zawodsky
2011-Aug-14 22:17 UTC
[asterisk-users] dynamically alter list of offered codecs (for faxing)
Hello everybody! Lately I've had experiences that I'd like to share with you: I did a some faxing over VOIP during the last two years. Not that much, lets say 1 fax per day on average. The setup is Old analog fax machine <-> Linksys PAP2 ATA <-> Asterisk 1.2 <-> DSL <-> VoIP Provider .... I would estimate that 80% of the faxes went through on the first try. The rest aborted transmission with some communications error and needed a second (or rarely a 3rd) try. Then suddenly, faxing didn't work that way any more. Every single fax needed many retries until it eventually went through. Now, since I didn't change anything on my side I wondered what had happened. I enabled sip debug on the CLI an made a test fax. I saw that my VoIP provider only offered codec alaw while the ATA was configured to only use ulaw. So I assume that Asterisk had to perform some transcoding and maybe that broke the reliability ... ??? Since I didn't change anything on my side, the only reason I can think of ist that my provider changed some hardware or whatever and suddenly offers only alaw. I reconfigured the ATA to only offer alaw and now every fax goes through on the first try without any problems. Through this experience I had an idea: The list of preferred codecs is "statically" set up in sip.conf. Is it possible to modify that list dynamically in the dialplan for the outbound leg? What I think of is to force the audio-stream to alaw f?r fax calls (= calls to/from a specific extension), but offer for example gsm for speech calls. (For example: FAX machine is connected to etension 1234. If a call is made from extension 1234, Asterisk should offer only alaw to the _provider_ side) Norbert