OK, tel me put my share of experience. I have a PRI TE212P which had onboard
VMP echo cancellation.
I am using Asterisk-1.2-18. the DTMF issue was really bad. It rang the
wrong extension some tings rang invalid extension. In the begining I didnt
knew that it was an DTMF issue.
Hi Stephane,
Solutions remove all the Asterisk modules as well as the addon modules, and
re-install Asterisk and Addon again.
# rm -rf /usr/lib/asterisk/modules/*
# rm -rf /usr/include/asterisk
Then reinstall it again.
It works for me.
Have you change the addons Makefile so that the ooh-323
Most of the users using this list do not experience the issue you are having,
rather than insult the admins, please trouble shoot and if you cannot, at least
post headers so others can.
-- Original Message --
From: Dimitri Volski [EMAIL PROTECTED]
Hi,
Yup, I agree. You have to post out your Zapata.conf for discussions, and
Have you tried out the cable terminations and also the incoming DTMF signal
voltage level?
It has so many possibilities for this case.
i am using tdm400P in my office. i tested that TDMF generated by
asterisk
Hi
Thanks for your help.
What do I need to change in Makefile for ooh-323 to be loaded?
Regards
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
clive.chan(Alpha Trilogies Networks)
Envoyé : mardi 10 juillet 2007 07:17
À :
Am Freitag, den 29.06.2007, 14:23 -0600 schrieb Anthony Francis:
Andres Paglayan wrote:
On Jun 29, 2007, at 12:50 PM, Lenz wrote:
Hello list,
I am getting the list with days of delay, take for example this
message:
As you can see, the message was posted on June 25th and was sent to my
Are you using SIP phones ? The issue can be from your phone to asterisk.
- Original Message -
From: AL Daei [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2007 8:42 AM
Subject: [asterisk-users] ZAP TDM and DTMF issue
Hi,
I'm curious if there is any
Hi
2007/7/10, Russell Bryant [EMAIL PROTECTED]:
As far as I know, the patch is ready for use. It has not yet been
merged into asterisk trunk, but I don't think there are technical
reasons for that. It's just a matter of someone else taking a final
look over it, and merging it in.
The
On 7/8/07, Dovid B [EMAIL PROTECTED] wrote:
What does the NexTone run for ?
- Original Message -
From: Andy Brezinsky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 03, 2007 8:17 PM
Subject: Re:
gincantalupo ha scritto:
Hi,
somtimes my Gigaset 450IP loses its registration.
Is there anybody who knows why and how to solve it?
TIA
Giorgio Incantalupo
I try some trick and i found:
maxexpiry=120
defaultexpiry=120
in sip.conf
I put this in a production env and all things are ok
No, please don't post headers. Headers tell us nothing. They don't tell
us such things such as DNS resolution problems, routing problems, if the
recipient's server is tempfailing, etc. The ONLY thing really useful
are the mail logs from the list server. The only people that have access
are Digium
On Tue, Jul 10, 2007 at 11:32:49AM +1000, Dimitri Volski wrote:
There is definitely something wrong with this list.
I have my emails sorted by date, and every day, the emails do not just
come on top, but get slotted in. Today (10 July 2007), I received about
6 emails from 29th of June,
Hi,
i'm having asterisk with sip working fine, including dundi lookups. The only
problem i'm having is that the dundi answer allways contains my internal,
private ip. Is there any way to set the targeting ip that is sent out in the
dundi answer (to my public ip or any other where i want to
Is it possible to share SIP phones registration information between two
different asterisk servers, that share the same realtime MySQL DB?
Regards,
Ricardo.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
On Wed, 27 Jun 2007 09:08:21 -0500, Matthew Fredrickson wrote:
You fixed your clocking then. That was what I was thinking of. Make
sure that your Dialogic card is also pulling timing from the Digium
card as well. What version of zaptel drivers are you running?
on a related issue, using
Hi Andreas,
In dundi.conf, look for the line of yours that is similar to this:
e164 = dundi-e164-canonical,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
Change ${IPADDR} to your external IP address or hostname, like so:
e164 = dundi-e164-canonical,0,IAX2,dundi:[EMAIL PROTECTED]
/${NUMBER}
Cheers,
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN BRIs for voice or data since early 90s.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent:
I know that asterisk does not support VAD. I poked around and saw some
reference to asterisk supporting CNG. Will CNG work without VAD ? If yes is
there any way to set this on asterisk 1.2.X for SIP using G729 ?
Thanks.
Dovid___
--Bandwidth and
Right... you dial *67 to block, however WE are the phone provider and
need to set the appearance value so that when our customers dial *67
we correctly block their caller-id from going out.
On 7/9/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Matt -
What do I need to do to set the outbound
I have 3 phones
P1 is a non video phone - grandstream
P2 is a Grandstream GXV3000
P3 is a Grandstream GXV3000
Using P1 to place a call to P2 I get audio only (as expected).
Then on P1 I transfer the call to P3 and I still only get audio.
At this point shouldn't the two video phones P2 and P3
Hi think that once SIP/SDP invite/reinvite is sent you can not change to
video stream.
On 7/10/07, Jerry Geis [EMAIL PROTECTED] wrote:
I have 3 phones
P1 is a non video phone - grandstream
P2 is a Grandstream GXV3000
P3 is a Grandstream GXV3000
Using P1 to place a call to P2 I get audio
No there is no SIP involved in this issue, its at IVR and ZAP incoming
channel.Are you using SIP phones ? The issue can be from your phone to
asterisk.- Original Message - From: AL Daei ar_daei at
hotmail.comTo: asterisk-users at lists.digium.comSent: Tuesday, July 10,
2007 8:42
I recorded some sound files using Asterisk record() app as ulaw file. I need to
edit these sound files. What kind of audio editor can I use to edit these files?
Gary Chen___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On Mon, Jul 09, 2007 at 11:42:19PM -0600, AL Daei wrote:
Hi,
I'm curious if there is any other option beside relaxdtmf in zapata , or any
where else to tune dtmf detection on TDM400 fxo boards.
in one of our sites provider is giving us 4 analog lines out of Adtran router
and Asterisk
What operating system are we talking ?
Darrell S. Long
BestWeb Corporation
Gary Chen wrote:
I recorded some sound files using Asterisk record() app as ulaw file.
I need to edit these sound files. What kind of audio editor can I use
to edit these files?
Gary Chen
Hi think that once SIP/SDP invite/reinvite is sent you can not change to
video stream.
On 7/10/07, Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/
// I have 3 phones
//
// P1 is a non video phone - grandstream
// P2 is a Grandstream GXV3000
sox
On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote:
I recorded some sound files using Asterisk record() app as ulaw file. I need
to edit these sound files. What kind of audio editor can I use to edit these
files?
Gary Chen
___
--Bandwidth and
The page has been wiki-fied and looks more usable, thank
Mat Kovach of NOOSS for the suggestion and enhancements.
http://nooss.org/wiki/Installing_Asterisk_From_Source
thnx,
-baji.
--
On 7/7/07, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my
Gary Chen wrote:
I recorded some sound files using Asterisk record() app as ulaw file. I need
to edit these sound files. What kind of audio editor can I use to edit these
files?
Gary Chen
You could try Audacity. They have both a windows and linux version.
Sox will convert them to a different format. If you want to edit them, you will need something more sophisticated than that. Audacity should be able to do it for most OS's.
Darrell S. Long
BestWeb Corporation
Andrew Latham wrote:
sox
On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote:
For interactive GUI-based editing, I have used audacity on linux
workstations.
I use command line sox for things such as format conversions.
Andrew Latham wrote:
sox
On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote:
I recorded some sound files using Asterisk record() app as ulaw file. I need
Hi kevin,
My problem it's i can't remove a caller from queue.
I defined a queue wich can accept only one call, and i have agent on
this queue,
my agent send a busy signal if he is on communication, and i want that
asterisk
remove new incoming call from the queue if the agent is busy (execute
On Tue, 2007-07-10 at 10:24 -0400, Gary Chen wrote:
I recorded some sound files using Asterisk record() app as ulaw file.
I need to edit these sound files. What kind of audio editor can I use
to edit these files?
You can use audacity, works on GNU/Linux and windows and is free
software (free
At 02:22 AM 7/10/2007, you wrote:
Considering that so Very Very few subscribers are having delays, there
is a 99% chance that you have something messed up on your side - DNS
reliability, your network, one or more of your MX servers, some goofy
anti-spam scheme, etc.
Or maybe it's just a VERY
Hi
sorry to bother, but I wasted a lot of time on this question, contact
several forum (as much english as french), and still no answer :-(.
In order to simulate an xfert, I thought that the PlayDTMF manager command
will be the right one.
All seems to be perfect :
- I have the sound
Hello all,
I'm working on a little project right now and have ran into a snag. Was
hoping someone would be kind enough to give me a few pointers to help me get
past the current issue...
I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...)
that I'm trying to get to
Hi Matt -
Right... you dial *67 to block, however WE are the phone provider and
need to set the appearance value so that when our customers dial *67
we correctly block their caller-id from going out.
Have you tried explicitly setting the CID variables to NULL strings?
- Noah
Jason Aarons (US) wrote:
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN BRIs for voice or data since early 90s.
I don't know what it's like in your area, but here, fractional PRI is
just not cost
Has anyone found a way to enable the g722 codec as a prefered codec in
the Polycom provisioning files for the 550's? I couldn't find a pref
for voice.codecPref.IP_550.
What needs to be put into the allow field (sip.conf) for asterisk to
allow the codec?
--
***
Forrest Beck
IAXTEL: 17002871718
No I haven't. Shouldn't I be able to set the appearance to like '4'
or '5', etc?
On 7/10/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Matt -
Right... you dial *67 to block, however WE are the phone provider and
need to set the appearance value so that when our customers dial *67
we
The Asterisk development team has released Asterisk version 1.2.21.1 and
1.4.7.1. These releases are minor updates to the releases that were
made yesterday to fix a couple of introduced issues. One issue was
related to the ODBC realtime driver. Another was related to music on hold.
Thank you
Quoting Benny Amorsen [EMAIL PROTECTED]:
SB == Stephen Bosch [EMAIL PROTECTED] writes:
SB Hi, folks: I remain intrigued by the gap in BRI implementation
SB between North America and Europe, and I wanted to get feedback
SB from the list members on the matter. I'm seriously considering
SB
Apologies if this has been brought up before, but extensive googling
and digging through my list archive didn't turn anything up.
Basically, I'm working on an AGI web app and need to read some digit
input. I'm having multiple issues with asterisk interpreting agi
commands at the moment,
Lee Jenkins wrote:
Arun Kumar wrote:
Hi
I already tried asterisk manager but Im not able to get status for each
queue member.
thanks
That must be a problem with your configuration. I get QueueMemberStatus
on my AMI interface (1.2):
Event: QueueMemberStatus
Privilege:
I am trying to use a macro to screen calls by calling several different
phones at the same time in different groups. Find me will not work and
queues will not work either. Trust me, I've tried them both and they
don't work like they should. Here is what I have:
A call comes into 6084 and
Noah Miller wrote:
The 430's have two line appearances. I'm trying to get the second line
registered to a different extension but for some reason it's not
allowing me to do this. The first line will register fine but the second
line never seems to register no matter how I swap the device ID's
It's not clear to me what happens when video and non-video clients are
using the same PBX together.
Let's say, from a video softphone I place a call through Asterisk,
through the analog phone line and my phone provider, to a regular analog
phone. Or the other way round - from an analog phone,
On 7/10/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Jason Aarons (US) wrote:
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN BRIs for voice or data since early 90s.
I don't know what it's like
Hi Matt -
No I haven't. Shouldn't I be able to set the appearance to like '4'
or '5', etc?
I can do that on the PRI's I've had experience with. I found that on
most landlines, this will show up as Unavailable or something
similar, but on most cell phones it will show that number.
- Noah
I will install Asterisk on my home server, I want to be able to route
video calls, but I need the Windows and Linux clients to be interoperable.
On Linux, it looks like Ekiga is a good candidate. But how about Windows?
Anyone using Kapanga in an Asterisk network that includes Ekiga? Are
these
Noah Miller wrote:
Hi Matt -
No I haven't. Shouldn't I be able to set the appearance to like '4'
or '5', etc?
I can do that on the PRI's I've had experience with. I found that on
most landlines, this will show up as Unavailable or something
similar, but on most cell phones it will
Hey Daniel,
I think adding the events would be a good idea.
Just open an issue on http://bugs.digium.com/ and attach your patch
there. Be sure to send a disclaimer to digium so your patch can be
included in the distribution (see
http://asterisk.org/developers/bug-guidelines for details).
Floyd wrote:
Hi everyone:
I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to
be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big
I am putting some scripts together to allow a local admin to add
extensions, then to reload the extensions, something like:
asterisk -r -x extensions reload
Are registered extensions forced to reauth?
Are active calls disrupted?
___
--Bandwidth
For windows you can't really go wrong with X-lite 3.0, the free version of
eye-beam.
http://www.counterpath.com/index.php?menu=Productssmenu=xlite
On 7/10/07, Florin Andrei [EMAIL PROTECTED] wrote:
I will install Asterisk on my home server, I want to be able to route
video calls, but I need
editor can I use to edit these
files?
Gary Chen
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Hi Florin -
On Linux, it looks like Ekiga is a good candidate. But how about Windows?
There's a windows version of Ekiga, too. It's at the very bottom of
the download page.
- Noah
___
--Bandwidth and Colocation Provided by
Hi Robert -
I am putting some scripts together to allow a local admin to add
extensions, then to reload the extensions, something like:
asterisk -r -x extensions reload
Are registered extensions forced to reauth?
Nope.
Are active calls disrupted?
Nope.
Reloads are safe to do in the
Hello,
I would like people to use soft phone, but they are used to have the standard
phone handset in their hands... Is there a USB handset or a handset that
connects to the audio card?
Thanks! __Yehavi:
___
From: Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Date: Wed, 11 Jul 2007 7:37 +0300
Hello,
I would like people to use soft phone, but they are used to have the standard
phone handset in
Hi Yehavi
You should consider USB Phone just as any sound device that is added to your
systems. Once the configuration has been done, it should appear as audio
device. Then you can configure your soft phone (ie Idefisk or any) to use
the new audio device as input, output and may be ringing
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