We've bought the G729 codec for lowering SIP bandwidth usage (we're
using grandstream phones) and we're quite happy with it up until I tried
using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations.
Weirdly enough, calls from IAXphone to the GS phone work just fine.
So are calls from both phones to voicemail. And from both phones to
analog phones connected to FXS ports.
Calls from GS to IAXphone ring, and once I answer the call in IAXphone,
I hear a very load noise.
Asterisk CLI shows this:
channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/205/1
of format GSM since our native format has changed to G729A
(not just once, over and over and over again till I hang up)
my sip.conf entry for the grandstream phone shows
disallow=all
allow=g729
and
reinvite=no
I did 'iax2 show channels' and 'sip show channels'
When I call from IAXPhone to GS, the IAX2 channel shows codec GSM and
the Sip channel shows codec G729A
When I call the other way around, Sip shows G729A and IAX2 shows GSM.
Hmm, seems ok...
I tried changing my sip conf to include allow=g729,gsm
Now the calls sounds fine, but the bandwidth is uses is near 20K instead
of just 6K (both phones are near me, and the Asterisk server is at a
remote location, and I can monitor bandwidth usage in my FW).
Can anyone help?
Thanks.
Shoval Tomer,
IT Manager,
SofTov Advanced Systems, Ltd.
Office: +972-3-9230686 ext. 179
Fax: +972-3-9216642
Mobile: +972-54-8000200
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