These show that a proper bridging tech module cannot be found to run
ConfBridge.
The debug message showing that a capability for ulaw couldn't be found was a
buggy
debug message which has now been fixed (it isn't a codec capability that
can't be found,
but a bridge capability). You need
Attach a debug[1] log so we can see what is happening.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
debug logs below:
Asterisk 1.8.4: http://pastebin.com/DFnKgSse
Asterisk trunk r319661: http://pastebin.com/B19tdbxJ
--
Hi,
I am trying to use ConfBridge application, but it throws Failed to
find a bridge technology to satisfy capabilities 0x4 (ulaw) error.
Please see console output below.
-- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005,
1001) in new stack
[May 19 13:36:05] DEBUG[7452]:
What version of Asterisk are you using? ConfBridge was rewritten in
trunk and would be good to see if you have the same issue.
Hi Paul,
I am using 1.8.4. Just tried with the latest trunk (SVN-trunk-r319661)
and it still doesn't work, this time throwing error as below:
-- Executing
Sorry, I wasn't maybe precise in my question. What I am looking for is
to use custom cadences (as defined in indications.conf) for ring tone
generated by 'r' option in a Dial command. Just found this patch:
https://issues.asterisk.org/view.php?id=14504
which does exactly this:
Hi,
Just noticed Asterisk is not playing 'ring' tone as defined in
indications.conf when Dial command is used with 'r' option.
For example:
[test]
exten = 123,1,PlayTones(ring)
exten = 123,n,Wait(5)
exten = 123,n,Playback(demo-congrats)
exten = 123,n,Hangup()
exten =
Hi,
I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer.
My extensions.conf:
[globals]
__TRANSFER_CONTEXT = transfer
[common]
exten = 123,1,Playback(demo-congrats)
exten = 123,n,Hangup()
exten = _0X.,1,Dial(SIP/${ext...@pstn-gw,60)
exten = _0X.,n,Hangup()
exten =
.
Regards,
Chris
2009/5/22 Martin asteriskl...@callthem.info:
it should work just fine; do you have the GSM codec compiled/loaded
core show modules like codec_gsm ... ?
OR that particular version has a BUG...
Martin
On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote
318
it seems that you doesent specify valid conference number
can you post meetme.conf
regards
Dhaval
On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote:
Hi,
I am not sure if I am doing something wrong, but I can't get MeetMe to
work with GSM codec (Asterisk
...@gmail.com:
On an entirely unrelated note, do you have the gsm asterisk sounds
installed? Maybe that vm-*.slin files don’t exist.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Maciejewski
Sent: Friday
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
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 kpflem...@digium.com:
Chris Maciejewski wrote:
Found unknown media
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 st...@geekinter.net:
On 22 May 2009, at 16:55, Chris Maciejewski wrote:
Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
audio
' (language 'en')
-- Hungup 'DAHDI/pseudo-1131226973'
2009/5/22 Kevin P. Fleming kpflem...@digium.com:
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
Hi,
I am not sure if I am doing something wrong, but I can't get MeetMe to
work with GSM codec (Asterisk 1.6.1 SVN r190371).
My config files below:
sip.conf:
[general]
context=common
canreinvite=no
bindport=5060
bindaddr=78.105.1.127
disallow=all
allow=alaw
allow=gsm
rtptimeout=600
Hi,
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
For example:
[default]
exten = _X.,1,Dial(SIP/u...@domain,30,M(GetOtherPartyInfo))
exten =
by Asterisk).
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski ch...@wima.co.uk wrote:
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside
-- INVITE [2] -- Phone 2
--- 200
OK [3] ---
What I want to do is capture Server header in 200 OK reply
generated by Phone 2.
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski ch
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like:
[default]
exten = _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/u...@domain.net,30,M(getCalledInfo))
2009/5/17 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
On Sunday 17 May 2009 09:10:47 Chris Maciejewski wrote:
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like
Maybe it is something to do with AGI - Dial command.
IFAIK you can't control Dial via AGI script.
From http://www.voip-info.org/wiki/view/Asterisk+AGI :
Dialing out
If the AGI application dials outward by executing Dial, control over
the call returns to the dialplan and the script loses contact
Hi,
I am trying to send 404 Not found reply, without any luck with the
following:
exten = 555,1,Playback(you-dialed-wrong-number,noanswer)
exten = 555,n,Playback(check-number-dial-again,noanswer)
exten = 555,n,Congestion()
However the above results in 500 Service Unavailable being send out.
Thank you all for help!
What I was trying to achieve was:
UA Asterisk
- INVITE -
--- 100 Trying --
183 Sess. Prog (sdp) -
[ here we play You dialled wrong... ]
-- 404 Not found -
And all is
Yes, 'causecode' parameter of Hangup application was missing at:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Hangup
I have added 'causecode' to the above wiki page now.
Thanks for your help,
Chris
2009/4/16 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
On Thursday 16 April 2009
Hi,
Is there any way to tell Asterisk not to generate additional headers like:
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
I can't find any relevant option in sip.conf file :-(
Thanks for help.
Chris
___
-- Bandwidth and
to
something else.
2009/3/15 Olivier oza-4...@myamail.com:
2009/3/15 Chris Maciejewski ch...@wima.co.uk
Hi,
Is there any way to tell Asterisk not to generate additional headers like:
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
I can't find any relevant option
Hi,
Is it possible to get information about SIP destination channel (created
after Dial command) somehow?
For example I would like to know what codec was used. I can do this for
originating channel with:
${CHANNEL(audionativeformat)}
but not sure how to do the same for destination channel?
Hi,
One of the solutions would be to overwrite standard *8 behaviour with
your custom macro that will 1) pickup a call as usual b) send
notification via AMI or whatever else you want. This can be done with
[applicationmap] in features.conf - see
Hi,
You can find some info about differences between 1.4 and 1.6 here:
http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?view=markup
Kind regards,
Chris
2008/8/28 --[ UxBoD ]-- [EMAIL PROTECTED]:
Hi,
I would like to give 1.6 a try and was wondering about the configuration
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