Stephen R. Besch
2003-Nov-11 12:57 UTC
[Asterisk-Users] Codecs and call failure with Grandstream
I know that this issue has been discussed a lot on this list in regard to some of the recent CVS's. However, it has come up as an issue on an older release (CVS Aug 05, 2003) as well. I thought that a heads up was in keeping with the philosophy of the list. Here are the details: Call from GS via * to remote IAX to PSTN. Sound stream is established from PSTN to GS but no sound from GS to PSTN. By the way, calls from GS to the PSTN via * worked correctly. Only the IAX bridge failed. It turned out to be a codec problem. The fix is the same as well. Add to sip.conf [general] (or on a phone by phone basis): disallow=all allow=alaw allow=ulaw You may also need to enable additional codecs. Stephen R. Besch