[May 22 10:29:34] DEBUG[6071]: chan_sip.c:4222 do_setnat: Setting NAT
on RTP to Off
Found RTP audio format 102
Found RTP audio format 101
Peer audio RTP is at port 78.105.1.131:8002
Found unknown media description format G726-16 for ID 102
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x0
(nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0
(nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
[May 22 10:29:34] NOTICE[6071]: chan_sip.c:7495 process_sdp: No
compatible codecs, not accepting this offer!
And asterisk is replying with "488 Not acceptable here"
Any help and suggestions very much appreciated.
Regards,
Chris
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Posted: Fri May 22, 2009 2:29 pm Post subject: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote:
Quote:
Found unknown media description format G726-16 for ID 102
It's right there.
Quote:
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
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org
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[May 22 16:48:20] DEBUG[6071]: chan_sip.c:4222 do_setnat: Setting NAT
on RTP to Off
Found RTP audio format 104
Found RTP audio format 101
Peer audio RTP is at port 78.105.1.131:8002
Found audio description format G726-32 for ID 104
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined -
0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
[May 22 16:48:20] NOTICE[6071]: chan_sip.c:7495 process_sdp: No
compatible codecs, not accepting this offer!
I note "Got unsupported a:fmtp in SDP offer"
from RFC 2327:
a=fmtp:<format> <format specific parameters>
This attribute allows parameters that are specific to a
particular format to be conveyed in a way that SDP doesn't have
to understand them. The format must be one of the formats
specified for the media. Format-specific parameters may be any
set of parameters required to be conveyed by SDP and given
unchanged to the media tool that will use this format.
It is a media attribute, and is not dependent on charset.
Is Twinkle sending this SDP incorrectly? Or some other issue?
Thanks
Chris
2009/5/22 Kevin P. Fleming <kpfleming@digium.com>:
Quote:
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
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org
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'us' does not include g726, so you have not configured your SIP
user/peer to support G.726.
Quote:
I note "Got unsupported a:fmtp in SDP offer"
No, that is not relevant. Asterisk's SDP parser does not pay much
attention to a:fmtp entries at this time.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org
_______________________________________________
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Posted: Fri May 22, 2009 3:41 pm Post subject: [asterisk-users] Can't get G.726 to work.
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 problem, as I don't have audio files for G726?
Will try converting .pcm to .g726 and see if that will fix MeetMe issue.
Regards,
Chris
2009/5/22 Steve Howes <steve@geekinter.net>:
Quote:
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
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Posted: Fri May 22, 2009 3:48 pm Post subject: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote:
Quote:
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 problem, as I don't have audio files for G726?
Will try converting .pcm to .g726 and see if that will fix MeetMe issue.
If you have codec_g726 loaded, you should be able to use prompt files in
any format that Asterisk can transcode from/to. 'core show translations'
should show you what formats Asterisk can convert to and from G.726.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org
_______________________________________________
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Posted: Fri May 22, 2009 4:14 pm Post subject: [asterisk-users] Can't get G.726 to work.
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:
-- <SIP/OpenSER-08208098> Playing 'entering-conf-number.slin'
(language 'en')
[May 22 18:07:04] WARNING[16881]: app_playback.c:447 playback_exec:
ast_streamfile failed on SIP/OpenSER-08208098 for entering-conf-number
-- Executing [501@services:7] SayNumber("SIP/OpenSER-08208098",
"1") in new stack
-- <SIP/OpenSER-08208098> Playing 'digits/1.slin' (language 'en')
-- Executing [501@services:8] Wait("SIP/OpenSER-08208098", "1") in new stack
-- Executing [501@services:9] MeetMe("SIP/OpenSER-08208098",
"11,MI") in new stack
== Parsing '/etc/asterisk/meetme.conf': == Found
-- Created MeetMe conference 1023 for conference '11'
-- <SIP/OpenSER-08208098> Playing 'vm-rec-name.slin' (language 'en')
-- Hungup 'DAHDI/pseudo-1131226973'
2009/5/22 Kevin P. Fleming <kpfleming@digium.com>:
Quote:
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 files.
> I suspect now it is the same problem, as I don't have audio files for G726?
>
> Will try converting .pcm to .g726 and see if that will fix MeetMe issue.
If you have codec_g726 loaded, you should be able to use prompt files in
any format that Asterisk can transcode from/to. 'core show translations'
should show you what formats Asterisk can convert to and from G.726.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org
_______________________________________________
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Posted: Fri May 22, 2009 9:20 pm Post subject: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote:
Quote:
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:
-- <SIP/OpenSER-08208098> Playing 'entering-conf-number.slin'
(language 'en')
[May 22 18:07:04] WARNING[16881]: app_playback.c:447 playback_exec:
ast_streamfile failed on SIP/OpenSER-08208098 for entering-conf-number
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.<foo>' where <foo> is the possible formats
that Asterisk could transcode from. Filenames specified to Playback
should not include the format extension.
Quote:
-- Executing [501@services:7] SayNumber("SIP/OpenSER-08208098",
"1") in new stack
-- <SIP/OpenSER-08208098> Playing 'digits/1.slin' (language 'en')
This did not fail. The .slin extension was added by ast_streamfile after
it found the correct format to play for this channel.
Quote:
-- Executing [501@services:8] Wait("SIP/OpenSER-08208098", "1") in new stack
-- Executing [501@services:9] MeetMe("SIP/OpenSER-08208098",
"11,MI") in new stack
== Parsing '/etc/asterisk/meetme.conf': == Found
-- Created MeetMe conference 1023 for conference '11'
-- <SIP/OpenSER-08208098> Playing 'vm-rec-name.slin' (language 'en')
Again, this did not fail.
Quote:
-- Hungup 'DAHDI/pseudo-1131226973'
The only failure of any kind that I see in this log is the call to Playback.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming@digium.com
Check us out at www.digium.com & www.asterisk.org
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Posted: Fri May 22, 2009 9:49 pm Post subject: [asterisk-users] Can't get G.726 to work.
On Fri, 22 May 2009, Kevin P. Fleming wrote:
Quote:
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.<foo>' where <foo> is the possible formats
that Asterisk could transcode from. Filenames specified to Playback
should not include the format extension.
The number of times I and others have "forgotten" that Asterisk chooses
the file based on codec and file type availability automagically makes me
wonder if it makes sense to change this so that if the file type (from the
list of known file types based on the formats loaded) is specified, use
it. Otherwise, proceed with the current code.
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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