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
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
http://www.voip-info.org/wiki-Asterisk+config+featur
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 C
ason, or maybe it is related to
something else.
2009/3/15 Olivier :
>
>
> 2009/3/15 Chris Maciejewski
>>
>> Hi,
>>
>> Is there any way to tell Asterisk not to generate additional headers like:
>>
>> X-Asterisk-HangupCause: Normal Clearing
>> X-As
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 ou
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 al
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 :
> On Thursday 16 April 2009 10:28:38 ContactTel Business wrote:
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 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 => _X.,
K response to INVITE
generated by Asterisk).
2009/5/17 David Backeberg :
> On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski 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)),
>> a
eServer v2
My scenario:
Phone 1 - INVITE [1] -> Asterisk --> INVITE [2] --> Phone 2
<--- 200
OK [3] ---
What I want to do is capture "Server" header in "200 OK" reply
generated by Phone 2.
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))
e
2009/5/17 Tilghman Lesher :
> 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
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
rtp
(G.726 RFC3551)
4096 (1 << 12) (0x1000) audio g722 (G722)
I will open a bug report.
Regards,
Chris
2009/5/22 Martin :
> it should work just fine; do you have the GSM codec compiled/loaded
>
> core show modules like codec_gsm ... ?
>
> OR that particul
isn't a valid
> conference
>
> its on line number 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 wrote:
>>
>> Hi,
>&
ely 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:
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
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 :
> Chris Maciejewski wrote:
>
>> Found unknown media description format G726-1
t 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 :
>
> On 22 May 2009, at 16:55, Chris Maciejewski wrote:
>> Capabilities: us - 0x100f (g723|gsm|ulaw|al
erence 1023 for conference '11'
-- Playing 'vm-rec-name.slin' (language 'en')
-- Hungup 'DAHDI/pseudo-1131226973'
2009/5/22 Kevin P. Fleming :
> Chris Maciejewski wrote:
>> Yes, I was missing "allow=g726" for this peer :-(
>>
>&g
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 =
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?
Any
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]: app
> 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 [501@
> 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
--
> 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
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 => 321,1,Dail(LOCAL/1...@
ial(SIP/1234,60,r(myring))
Thanks
Chris
2009/6/13 David Backeberg :
> On Sat, Jun 13, 2009 at 11:27 AM, Chris Maciejewski wrote:
>> Hi,
>>
>> Just noticed Asterisk is not playing 'ring' tone as defined in
>> indications.conf when Dial command is used with &#x
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