Hi,
I have just installed Asterisk 1.4.1 on my machine… and all I have done is added a new SIP user:
[101]
username=101
type=friend
secret=101
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
host=dynamic
dtmfmod=rfc2833
context=default
callerid=Antonio Broughton <101>
I then connected to the Asterisk server using an snom 360 softphone (that works)… but… when I tried to dial 1000, I get the following at the console:
-- Executing [s@default:5] BackGround("SIP/101-081c32f8", "demo-congrats") in new stack
-- <SIP/101-081c32f8> Playing 'demo-congrats' (language 'en')
[Mar 6 16:38:06] WARNING[26394]: codec_gsm.c:144 gsmtolin_framein: Invalid GSM data (1)
[Mar 6 16:38:06] WARNING[26394]: translate.c:197 framein: gsmtolin did not update samples 0
[Mar 6 16:38:06] WARNING[26394]: translate.c:163 framein: no samples for gsmtolin
[Mar 6 16:38:06] WARNING[26394]: translate.c:163 framein: no samples for gsmtolin
[Mar 6 16:38:06] WARNING[26394]: translate.c:163 framein: no samples for gsmtolin
It just repeats that until I end the call… the call starts off fine, but then starts to skip… (possibly because of the “warning” messages)?
Does anyone have an idea why this is occuring?
A google search of “framein: no samples for gsmtolin” returns a page in German (which I don’t speak), but it has something about Asterisk 1.4.1…