just post your peer configs for one of your clients that don't show on the
log.
mostly it's IAX peers that don't show on the logs if not configured to.
All my clients are sip peers actually.
Here is the users.conf entry for one of the users that doesnt show on logs.
[8006]
username = 8006
Anyone have used one of the new Cisco SPA525G with Asterisk ? Will be
reading manual before starting to play with, but would really appreciate if
you could share some tips with me. Thanks
G.
___
-- Bandwidth and Colocation Provided by
Steve Underwood wrote:
Hi,
If anyone is interested in the low speed modems needed for POS
applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I
had some spare time while travelling, and finally got the V.22bis code I
started a long time ago into a start where its
Thanks very much.
By the way, what exactly does silence threshold mean, how does it work, and
what does the threshold value represent (bitrate? integer?)? The amd.conf
and voip-info wiki doesn't describe it.
On Tue, Apr 28, 2009 at 5:58 PM, Matt Riddell li...@venturevoip.com wrote:
On
Hi,
Is there some more thorough documentation of this change that has
happened in 1.6? The upgrade.txt and changes.txt files mention it, but
I have already seen details of this change that do not appear to be
documented except in conversations on the mailing list...
1) It appears that it is no
Hello,
I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in
registering users.
As i see from debug it successfully reads from users.conf but later,when a
user tries to logon it say peer not found
And there were an error msg about mysql about the username field..Smthing
changed
Hi,
I have some files in mp3 in my Asterisk but when I play it the volume is lo=
wer than wav files. Both the files (wav and mp3) are encoded with the same =
amplitude. In anothers players the audio volume of these files are equal.
Can I fix this diference between volume of mp3 and wav file?
Check this link.
http://www.voip-info.org/wiki/view/Asterisk+mpg123+faking+it
Are you using mpg123? What version of Asterisk are you using? Why dont
you just SOX the files into .gsm, .wav or some other Asterisk-happy
format?
_
From: asterisk-users-boun...@lists.digium.com
Hi,
Thanks for your reply.
We donot kept any absolute time out's.
And we have remove the AMD and kept only the play back,
it works fine.
Any help is highly appreciated.
Thanks.
On Wed, Apr 29, 2009 at 6:35 AM, Matt Riddell li...@venturevoip.com wrote:
On 28/04/2009 4:56 p.m., Sam Hawkin
On Wednesday 29 April 2009 07:27:44 Steve Davies wrote:
Hi,
Is there some more thorough documentation of this change that has
happened in 1.6? The upgrade.txt and changes.txt files mention it, but
I have already seen details of this change that do not appear to be
documented except in
I'm using Asterisk 1.4.19 and asterisk-addons 1.4.7.
So to play mp3 it uses format_mp3.so module.
I use mp3 because I need play the same audio file in the web.
--- Em qua, 29/4/09, Danny Nicholas da...@debsinc.com escreveu:
De: Danny Nicholas da...@debsinc.com
Assunto: Re: [asterisk-users]
good point :)
From: abalas...@evaristesys.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Wednesday, April 29, 2009 1:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] I
Check this page out.
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
You might have to do some kind of custom-app tweak to play the mp3 at an
increased volume.
In my installation, when I take the
2009/4/29 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
Let's also be clear about what Gosub is replacing. Gosub replaces Macro for
AEL2. The side effects of this are relatively unfelt, unless you're doing
something unusual like defining subroutines in AEL and calling them from
I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO
modules.
When I plug one PSTN-line into a FXO-port I am able to receive calls on
this line and I can also make calls from an internal SIP-phone to the
external PSTN-network.
Still I am bothered about something that appears on the
Hi
I am using asterisk-1.6.0.6 and I have noticed strange behaviour
lately. When a user ends his call asterisk executes twice the h
extensions (in my case this is the AGI script) and writes this to the
logs:
cdr.c: CDR on channel 'SIP/xx-b6623038' already posted.
and after that it crashes
On Wednesday 29 April 2009 11:45:13 Steve Davies wrote:
2009/4/29 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
Let's also be clear about what Gosub is replacing. Gosub replaces Macro
for AEL2. The side effects of this are relatively unfelt, unless you're
doing something unusual like
A couple of days ago we decide to test 1.6 branch, we are using 1.4
in our production environments, and to my surprise, when ported parts of
our dialplan, I found that any macro declaration in the AEL dialplan, we
use them a lot for subroutines, was ported as gosub when compiled.
We are
Wrong list. asterisk-dev is for changing the C source code of Asterisk.
That's part of why you didn't get a response yesterday.
On Wed, 29 Apr 2009, Juan Miguel Quiros Arrieta wrote:
I have to develop an application using the VeriFone vx510 device and I
read this device needed or could use a
Wrong list. asterisk-dev is for changing the C source code of Asterisk. I
don't think AGI's count or are considered for inclusion into the
subversion repository as stated by one of your conditions for payment.
On Wed, 29 Apr 2009, Alistair Cunningham wrote:
I'd like to offer a bounty for a
On Wed, Apr 8, 2009 at 10:23 AM, Shaun Ruffell sruff...@digium.com wrote:
David Backeberg wrote:
Hello there:
I think I have a silly kernel configuration problem. I'm running:
* vanilla 2.6.27.10 kernel built from source
* dahdi-2.1.0.4 built from source
So far so good,
dahdi module loads
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Andrew Nowrot wrote:
This had happened twice so far. Does anyone know what is causing this.?
Start by upgrading to 1.6.0.9, then if it continues you can start
tracking it down.
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
For some reason, I have been unable to find the answer to this online or in
books...
I want to have a click-to-connect feature on my website where the user enters
their phone number and then my asterisk server calls their phone and my phone
and connects the two calls to each other.
All I
try adding callerid=CIDNAME CIDNUM
this will force your callerID in your DIalplan
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Wed, 29 Apr 2009 09:38:58 +0300
From: oguzh...@bilkent.edu.tr
To:
On Wednesday 29 April 2009 14:14:58 don rhummy wrote:
For some reason, I have been unable to find the answer to this online or in
books...
I want to have a click-to-connect feature on my website where the user
enters their phone number and then my asterisk server calls their phone and
my
On Wed, Apr 29, 2009 at 07:41:14PM +0200, jonas kellens wrote:
I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO
modules.
When I plug one PSTN-line into a FXO-port I am able to receive calls on
this line and I can also make calls from an internal SIP-phone to the
external
Thank you .. appreciated.
Best Regards,
--
SplatNIX IT Services :: Innovation through collaboration
- Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 28 April 2009 15:35:14 --[ UxBoD ]-- wrote:
HI,
I am trying to setup CDR with ODBC and MySQL but get the
On Apr 28, 2009, at 11:36 PM, Gondar Monn wrote:
Anyone have used one of the new Cisco SPA525G with Asterisk ? Will
be reading manual before starting to play with, but would really
appreciate if you could share some tips with me. Thanks
We tested one a few months ago. They work like
OK I will do that. i let you know about the results.
Cheers
On Wed, Apr 29, 2009 at 9:21 PM, Barry L. Kline blkl...@attglobal.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Andrew Nowrot wrote:
This had happened twice so far. Does anyone know what is causing this.?
Start by
Okay, I can't find what might be causing this. Here is what I got:
Asterisk server hooked up to a digital T1 line (full 24-channel) via a
Digium card.
Verizon has turned on caller ID on the first line (I can guarantee it
is on as I can hear the FSK tones on this line but not the others).
On 30/04/2009 2:25 a.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
We donot kept any absolute time out's.
And we have remove the AMD and kept only the play back,
it works fine.
Any help is highly appreciated.
Ok, so when you remove AMD and keep playback, how long is the message.
On 30/04/2009 5:41 a.m., jonas kellens wrote:
I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules.
[*Apr 29 16:26:53] WARNING[8927]: chan_dahdi.c:11951 process_dahdi:
Ignoring any changes to 'signalling' (on reload) *
Changes to signalling are not used when you reload -
On 29/04/2009 9:06 p.m., Roi Stork wrote:
Thanks very much.
By the way, what exactly does silence threshold mean, how does it work,
and what does the threshold value represent (bitrate? integer?)? The
amd.conf and voip-info wiki doesn't describe it.
Basically when you speak, silence
On 30/04/2009 7:14 a.m., don rhummy wrote:
For some reason, I have been unable to find the answer to this online or in
books...
I want to have a click-to-connect feature on my website where the user
enters their phone number and then my asterisk server calls their phone and
my phone and
Does anybody know of a way to make another parking lot for version 1.2? We
have a multi-tenant setup and it is set for x700 for parking. Well we added
some new users and not thinking, we assigned them x700. I can't change the
parking number as it will mess up the other users and the new user
Um, no... I want to implement it myself. My question is with regard to
asterisk, and getting it top actually make calls, etc, what do i need to make
those outgoing and connecting calls with asterisk?
--- On Wed, 4/29/09, Matt Riddell li...@venturevoip.com wrote:
From: Matt Riddell
On 30/04/2009 11:00 a.m., don rhummy wrote:
Um, no... I want to implement it myself. My question is with regard to
asterisk, and getting it top actually make calls, etc, what do i need to make
those outgoing and connecting calls with asterisk?
Did you go to the link I sent you?
It was to a
You're saying this is worth $5k ? This can be done in 2-3 hrs so are
you really charging
$1666-2500 an hour ?
Martin
On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards
asterisk@sedwards.com wrote:
If anyone would like to write this, and it gets accepted into the
Asterisk subversion repository
Un-top-posting...
On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 29 Apr 2009, Alistair Cunningham wrote:
If anyone would like to write this, and it gets accepted into the
Asterisk subversion repository for a future Asterisk version,
Integrics is
I just installed asterisk 1.6.0.9 in the hope of using the d option in
ExtenSpy(). I found this on the internet:
- w - Enable 'whisper' mode, so the spying channel can
talk to
- the spied-on channel.
- W - Enable 'private whisper' mode, so
automon is not working for me with asterisk 1.4.22.1
in extension.conf
[globals]
DYNAMIC_FEATURES=automon
dial is with w
feature.conf
automon = *1
-- Executing [...@internal:1] Playback(SIP/218-007556b0, transfer) in new
stack
-- SIP/218-007556b0 Playing 'transfer' (language 'en')
--
No more questions. This all can be done in 2-3 hrs [PERIOD].
Martin
On Wed, Apr 29, 2009 at 8:02 PM, Steve Edwards
asterisk@sedwards.com wrote:
Un-top-posting...
On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 29 Apr 2009, Alistair Cunningham
I found a version of valetparking.c that looks like what I need for adding more
parking. However when I try and compile it in 1.2, it is looking for a file
parking.h, which isn't in my install. I even looked through CVS and 1.0 and
1.2 to make sure it isn't an old file and I can't find it
Hi,
Thanks for your reply.
We I remove the AMD it plays the message in the 12 seconds.
It takes 16 seconds before AMD disconnects.
We are using Asterisk 1.2.4
Any help is highly appreciated.
Thanks.
On Thu, Apr 30, 2009 at 3:00 AM, Matt Riddell li...@venturevoip.com wrote:
On 30/04/2009
On 30/04/2009 4:26 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
We I remove the AMD it plays the message in the 12 seconds.
It takes 16 seconds before AMD disconnects.
We are using Asterisk 1.2.4
Any help is highly appreciated.
Few things:
1. Play the message twice without AMD
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