On Fri, 22 May 2009, Kevin P. Fleming wrote:
> This is not MeetMe, it's Playback. You specified a filename with '.slin'
> in it to Playback, so then Asterisk attempts to find a filename called
> 'entering-conf-number.slin.' where is the possible formats
> that Asterisk could transcode from. Fi
Chris Maciejewski wrote:
> I do have codec_g726 loaded. As I mentioned before
> Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite
> there is only fpm-sunshine.wav file. It is only MeetMe which is not
> working:
>
> -- Playing 'entering-conf-number.slin'
> (language 'en')
I do have codec_g726 loaded. As I mentioned before
Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite
there is only fpm-sunshine.wav file. It is only MeetMe which is not
working:
-- Playing 'entering-conf-number.slin'
(language 'en')
[May 22 18:07:04] WARNING[16881]: app_p
Chris Maciejewski wrote:
> Yes, I was missing "allow=g726" for this peer :-(
>
> Playback(/var/lib/asterisk/moh/fpm-sunshine)
>
> works OK now, however I still can't get MeetMe to work.
>
> Before I had similar problem, when MeetMe wasn't working with GSM
> codec because I was missing .gsm audio
Yes, I was missing "allow=g726" for this peer :-(
Playback(/var/lib/asterisk/moh/fpm-sunshine)
works OK now, however I still can't get MeetMe to work.
Before I had similar problem, when MeetMe wasn't working with GSM
codec because I was missing .gsm audio files.
I suspect now it is the same prob
Chris Maciejewski wrote:
> Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
> audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined -
> 0x0 (nothing)
'us' does not include g726, so you have not configured your SIP
user/peer to support G.726.
> I note "Got unsupported a:f
On 22 May 2009, at 16:55, Chris Maciejewski wrote:
> Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
> audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined -
> 0x0 (nothing)
Codec not enabled on that peer?
S
___
-- Bandwidth
Hi Kevin,
Thanks for your reply. I switched to G726 32Kbps but still no luck:
INVITE
[SIP headers omitted]
v=0
o=1 1291673978 653998617 IN IP4 192.168.7.55
s=-
c=IN IP4 78.105.1.131
t=0 0
m=audio 8002 RTP/AVP 104 101
a=rtpmap:104 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101
Chris Maciejewski wrote:
> Found unknown media description format G726-16 for ID 102
It's right there.
> And asterisk is replying with "488 Not acceptable here"
Asterisk does not support G726-16, it only supports G726-32.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
44
Hi,
I have both codec_g726.so and format_g726.so loaded:
r...@test:~# asterisk -r -x "module show" | grep 726
codec_g726.so ITU G.726-32kbps G726 Transcoder 0
format_g726.so Raw G.726 (16/24/32/40kbps) data 0
But when I try to dial into Asterisk w
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