asterisk users - Jan 2005 - Codec mismatch between SIP (BT) and IAX Phone

Hi,

I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client
(IAXPhone):
- when I call from Iax to SIP sound works
- when I call from Sip to Iax sound doesn't work, I get :

Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping
incompatible voice frame on IAX2/200/1 of format gsm since our native format
has changed to ulaw

Why is Asterisk not satisfied with gsm packets - it should transcode if
necessary ?
I have enabled gsm and ulaw in both configs, but it seems not sufficient.

Any advice, help ?

Thanks in advance,

regards,

Rob.

In both configs there are only general codec settings .
I have in sip.conf (snippet):
[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
context = from-sip ; Send unknown SIP callers to this context

And in iax.conf (snippet) :
[general]
bindport = 4569           ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
;delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
jitterbuffer=yes
mailboxdetail=yes
authdebug=yes

> (IAXPhone):
I suppose you're talking about Steve Sokol's phone
If so, then this phone only support gsm.
> Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping
> incompatible voice frame on IAX2/200/1 of format gsm since our native
format
> has changed to ulaw
> 
> Why is Asterisk not satisfied with gsm packets - it should transcode if
> necessary ?
> I have enabled gsm and ulaw in both configs, but it seems not sufficient.
Yes, * will transcode, but you specified in the IAX Phone config that
you allow this one tu use gsm AND ulaw, so instead of transcoding, *
just tell the IAX Phone to switch to uLaw, since the originating party
sends it in ulaw.

Just change your iax.conf to only allow gsm on the IAX Phone like this :

disallow=all
allow=gsm