I’m unsure if what I’m seeing is correct or not in my channel info for a sip cleint making a call in terms of which codecs are being used. I am running Asterisk 1.4.2, I have sip.conf set to force alaw (g.711) on my sip clients.
When I call up using one of those clients and do a 'core show channel x/channel" I see this (excerpt):
NativeFormats: 0x8 (alaw)
WriteFormat: 0x40 (slin)
ReadFormat: 0x8 (alaw)
WriteTranscode: Yes
ReadTranscode: No
1st File Descriptor: 16
I’m a little confused because I would think the NativeFormats, WriteFormat and ReadFormat would all be alaw, instead of alaw, slin, alaw.
Is this expected? or do I have a configuration issue? I am battling some other sound quality problems for this system (it’s an inhouse only system, meaning no outside calls) so I’m going through each possible area one step at a time and looking for anomilies.
Thanks,
Colin