[asterisk-users] IAX trunks and audio codecs

Vieri rentorbuy at yahoo.com
Fri Apr 30 05:51:02 CDT 2010


Hi,

I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the "primary" Asterisk server where all the SIP phone extensions are registered.

The IAX trunk settings are something like this (all servers have this identical except for the "host" field):

[inbound]
deny=all
allow=alaw
allow=gsm
type=friend
host=192.168.250.111
secret=inboundpass
auth=plaintext
requirecalltoken=no
qualify=yes
context=from-inbound
username=inbound
trunk=yes

I'm trying to force the use of alaw because some of the local SIP extensions use this codec (a minor percentage use gsm) and none use ulaw.
So I suppose that if the first Asterisk server that receives the call and sends it out to the main server via IAX encodes in alaw then the main server won't have to transcode if the destination is also alaw (most SIP phones).
This "should" save some CPU processing in the main Asterisk server, right?

So my trouble is with this message on the main Asterisk server when it receives a call from a secondary server via IAX:

Apr 30 12:19:59] NOTICE[14517] channel.c: Dropping incompatible voice frame on IAX2/inbound-2255 of format alaw since our native format has changed to 0x4 (ulaw)

Why is it changing to ulaw if I'm explicitly allowing only alaw and gsm and denying the rest?

Thanks,

Vieri



      



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