[Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

[Robert Rozman]

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