I run Asterisk 1.8 at home for my own amusement. I’ve just got a SIP trunk running again to SIPGate which had stopped running for some reason. I’d set the DID for the SIPGate number to check for faxes and then, if it’s not a fax, go to my MOH application, but for some reason it didn’t work.
I also couldn’t access voicemail from the SIPGate trunk either, but I could get it to work with a phone connected to an ATA.
I had a look in the logs:
channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
Since slin is, I think, the Asterisk native format this is quite bad. My assumption is that SIPGate sends through G.729 encoded audio regardless of whether you want it to or not (as configured in your allow/disallow lines for the SIP trunk PEER).
G729 from SIPGate works to the ATA because the ATA supports G729 and G729 to G729 pass-through works without any extra requirements from Asterisk.
So – in order to get SIPGate to Asterisk apps working again I installed the G729 codec binaries from here:
and restarted Asterisk. Now a:
core show translations
shows I can convert between G729 and loads of other codecs.