Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, Zap/3/8801234) in new stack
[Feb
On Mon, 25 Feb 2008, Ian wrote:
Mojo with Horan Company, LLC said the following on 22-Feb-08 07:58 PM:
Sorry, I jut got your other message stating the steps your boss' secretary
uses to transfer calls, so this question's time is past.
I'm curious if the 'flash' button is the only way
Hi!
I have 3 TDM800P with 8 incoming lines each.
Is it possible to detect DTMF-tones simultaniously on all 24 Zap-channels?
Or is there a limitation on this?
I guess dtmf-detection is a kind of DSP-operation, which consumes a lot of
CPU-power.
Will the zap-driver be able to sample all channels
21 feb 2008 kl. 14.38 skrev Mayur:
Hi,
I would like to configure asterisk to allow INVITE for hold to pass
through it and not provide music on hold by itself. Can anyone help
me out here?
That's a feature that doesn't exist in Asterisk today, but could
easily be added.
Regards,
/olle
Mike Hammett [EMAIL PROTECTED] writes:
*bump*
This is not some silly forum. *plonk*
/Benny
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Hi All;
I am using IAX Trunk and I used ddns (dyndns.org) with
the host (host=xyz.dyndns.org), when I restart the
router who has the hostname xyz.dyndns.org then its IP
address change and updated, but at asterisk level
still it keeps sending for the old IP address and
sometimes this problem does
It is, however, heavily trafficked and easy for someone to miss an email.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Benny Amorsen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 25, 2008 3:44
I thought it was odd, but I've had other devices work properly with that
information.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Just as a silly one, try 'asterisk -rx iax2 reload'. I'm not sure if it'll
work or not, but it should force a recheck of the hostname.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: 25 February 2008 02:21 PM
To:
Johansson Olle E wrote:
That's a feature that doesn't exist in Asterisk today, but could
easily be added.
Actually, it is there... setting 'mohinterpret' to passthrough will get
as close as Asterisk can get to 'proxy mode' for this purpose, but it
will solve the OP's question.
--
Kevin P.
Gordon Henderson said the following on 25-Feb-08 10:26 AM:
On Mon, 25 Feb 2008, Ian wrote:
Mojo with Horan Company, LLC said the following on 22-Feb-08 07:58 PM:
Sorry, I jut got your other message stating the steps your boss' secretary
uses to transfer calls, so this question's time
Dear Everyone,
I expect this page should delivery some hints for those who wish to dig into
various possibility of asterisk setup.
Here, I use asterisk phone-back features to have my web clients having a
free-trial-cal
to their designated mobile or landlines as desired.
How
Dear Everyone,
I expect this page should delivery some hints for those who wish to dig into
various possibility of asterisk setup.
Here, I use asterisk phone-back features to have my web clients having a
free-trial-cal
to their designated mobile or landlines as desired.
How
On Sat, 2008-02-23 at 07:52 +0200, Yehavi Bourvine +972-8-9489444 wrote:
The people here don't let me even try it as they are afraid it will
consume the
battery more than when it is used the usual way. Is this true?
Yes, is true.
You must have to disable the automatic wireless LAN scan
25 feb 2008 kl. 14.35 skrev Kevin P. Fleming:
Johansson Olle E wrote:
That's a feature that doesn't exist in Asterisk today, but could
easily be added.
Actually, it is there... setting 'mohinterpret' to passthrough will
get
as close as Asterisk can get to 'proxy mode' for this purpose,
Hi Asterisk users,
For those of you in the greater New York area BarCampNYC3 is on again
this year.
It's being held on the 15th and 16th of March at the Polytechnic
University in Brooklyn. http://barcamp.pbwiki.com/BarCampNYC3
As always entry is FREE but expect to come and present,
Which Area codes are you looking for? Is this a multiple Exchange. I use
Link2voip but they don't scale well unless you're running a hobby. The best
solution is to get your PRI T1 from a telco.
On 2/23/08, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I posted the same question on asterisk-biz
If I have an asterisk server and lets say 150 asterisk clients all
running SIP
and I wish to initiate a one way call out and bring those 150 units into
a one
way conference/ intercom group/ something ...
What is the fastest method?
how long would that call setup take before I can start talking?
I may get T1 PRI for my own city, but I can't get PRIs in all the cities
where I might have customers.
On Mon, Feb 25, 2008 at 9:55 AM, broadband Voice [EMAIL PROTECTED]
wrote:
Which Area codes are you looking for? Is this a multiple Exchange. I use
Link2voip but they don't scale well unless
On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Generally, the rule is that you can't remove any of the res_*
modules.
Thanks for the tip. At this point, I have the following in
modules.conf, but when I type reload, it still loads stuff I
disabled such as DunDI:
Hi,
How could i trace what codec my voip provider is using?
I still can't make the IVR for that certain provider, i still can't hear
the sound, but i can see it connecting. where else should i look?
Regards,
Ron
Steve Totaro wrote:
On Thu, Feb 21, 2008 at 12:40 PM, Ron [EMAIL PROTECTED]
On Monday 25 February 2008 10:04, Vincent wrote:
On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Generally, the rule is that you can't remove any of the res_*
modules.
Thanks for the tip. At this point, I have the following in
modules.conf, but when I type
On Mon, 25 Feb 2008, Tilghman Lesher wrote:
On Monday 25 February 2008 10:04, Vincent wrote:
On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Generally, the rule is that you can't remove any of the res_*
modules.
Thanks for the tip. At this point, I have the
Hi,
I recently installed a TE120P in my lab with a full voice PRI (23 channels +
1 D channel). Everything is working well except echo cancellation; for the
most part this isn't an issue unless one of the users is in a conference.
I'm getting the following error when a call is picked up (incoming
Dear All,
Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc.
and will be able to receive faxes and negotiate with voip CPE's like
ATA's to transmit faxes which comes from FXO cards to VoIP Devices using
T38 ? it is possible to compile this version of app_fax to work with
Hello all,
Our setup consists of an asterisk server (version 1.4.8 ) with a TDM400 card
(4 FXO ports) and about 10 aastra 5Xi phones (2 phones are 57i and the rest
55i). The asterisk server is also configured with two SIP trunks.
The problem that we're having is that if at any time there is an
Hi,
I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
I've ran into an issue. After a call placed any DTMF tone causes the server
to lock up entirely. Calls placed work just fine (except for a problem with
echo cancellation). The phone registered to the server is a Linksys
One of my ATAs became unreliable recently, and I replaced it with 2 ATAs
that I had spare (a PAP2, and a cisco 186).
A while back, I replaced an spa3k with a linecard (the spa3k now is only
used to connect the FXO and FXS ports during a power cut).
I am considering replacing the ATAs with
On Mon, Feb 25, 2008 at 05:32:24PM -0300, Fernando Berretta wrote:
Dear All,
Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax etc.
and will be able to receive faxes and negotiate with voip CPE's like
ATA's to transmit faxes which comes from FXO cards to VoIP Devices
On Mon, Feb 25, 2008 at 11:46:09AM -0600, Brett Crapser wrote:
On Mon, 25 Feb 2008, Tilghman Lesher wrote:
On Monday 25 February 2008 10:04, Vincent wrote:
On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Generally, the rule is that you can't remove any
On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote:
Hi,
I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
I've ran into an issue. After a call placed any DTMF tone causes the server
to lock up entirely. Calls placed work just fine (except for a problem with
echo
My guess is a mismatch between Asterisk, Zaptel, and libPRI. Make sure
you are running the latest of each.
Tzafrir Cohen wrote:
On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote:
Hi,
I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
I've ran into an issue. After
arkda wrote:
[Feb 25 12:54:01] WARNING[8661]: chan_zap.c:1437 zt_enable_ec: Unable to
enable echo cancellation on channel 1 (Argument list too long)
Can you tell us what versions of Asterisk and Zaptel you are using?
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The
Johansson Olle E wrote:
Interesting. I don't see the support for remote hold in chan_sip, but
I guess I still can be surprised by the things that hide in that
small, tiny and well-organized piece of code :-)
/me blushes
That was my mistake... the capability is in place, but chan_sip hasn't
It's the latest from each (branch, not trunk): Asterisk revision 104093,
zaptel revision 3849, libpri revision 529 all from svn.
On Mon, Feb 25, 2008 at 3:50 PM, Eric Wieling [EMAIL PROTECTED] wrote:
My guess is a mismatch between Asterisk, Zaptel, and libPRI. Make sure
you are running the
Nothing in the console aside from what I've posted. When a DTMF tone is
played the server freezes instantly, hard reboot required.
Distribution is SuSE 10.2, kernel 2.6.18.8-0.7-default
The actual dialplan on this server is very simple, only one phone and a few
Dial commands via SIP to another
Asterisk revision 104093, zaptel revision 3849, libpri revision 529 all from
svn.
OT from echo cancellation, but I've ran into some other issues as well
(listed in the other active thread, DTMF tone crashes server (Asterisk
1.4.18 with Digium TE120P)) with the server freezing when a DTMF tone is
Hello,
I am having a problem which stems from the fact that the Agents that
handle my call queues are unaware if there are are people waiting in the
queue. I have found the following configuration options but they are
not very helpful.
in queues.conf:
announce=xxx
The announce = XXX option
Aastra tech are a bit slow, be sure to put a ntp server into your LAN and
point Aastra's to it.
Your problems will be solved.
Adrià Vidal
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Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module and one fxo module. I am using kernel 2.6.24 and get the
following log entries:
[Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [EMAIL
Hi John,
John Covici wrote:
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module and one fxo module.
It looks like you might be hitting a regression with DTMF tone
generation in the latest zaptel
On 2/25/08, Vincent [EMAIL PROTECTED] wrote:
On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Generally, the rule is that you can't remove any of the res_*
modules.
Thanks for the tip. At this point, I have the following in
modules.conf, but when I type
How would I use that with kernel 2.6.24?
on Monday 02/25/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote
Hi John,
John Covici wrote:
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module and
John covici wrote:
How would I use that with kernel 2.6.24?
Ok...good point. Tonight I'll make a patch for issue 11855 to fix this,
even if it means removing the dtmf-twister code until I can figure out
what is going on as pat suggested.
___
--
There already is an ntp server on the LAN, but the phones still freeze.
On Mon, Feb 25, 2008 at 2:18 PM, Adrià Vidal [EMAIL PROTECTED] wrote:
Aastra tech are a bit slow, be sure to put a ntp server into your LAN and
point Aastra's to it.
Your problems will be solved.
Adrià Vidal
arkda wrote:
Asterisk revision 104093, zaptel revision 3849, libpri revision 529 all
from svn.
Revisions of what branches? We need to know the URL you checked out for
each of these, not just the revision number.
Also, have you confirmed in your kernel message log that the version of
Zaptel
Sorry, 1.4. Keep forgetting 1.2 is still around. These were built with:
svn co http://svn.digium.com/svn/libpri/branches/1.4/ libpri
svn co http://svn.digium.com/svn/zaptel/branches/1.4/ zaptel
svn co http://svn.digium.com/svn/asterisk/branches/1.4/ asterisk
There has never been another version
I have a simple asterisk install (1.4.18), and want to use call parking. I
can successfully park a call (I see on the CLI that the call is parked to
701). Everything is pretty default.
However, I can't pickup a call from another phone. When I dial 701 from a
phone, asterisk can't find that
On Mon, 2008-02-25 at 20:03 -0500, Michelle Dupuis wrote:
However, I can't pickup a call from another phone. When I dial 701 from a
phone, asterisk can't find that extensions and notifies the person picking
up that the extension doesn't exist. (It appears that asterisk is looking
for
Did you pay attention to the following bit?
--8snip
For simple dialplans first edit features.conf as desired, then put this
into your extensions.conf
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf:
include = parkedcalls
I have 2 contexts in my extensions.conf: internal and external calls. I
have included the parkedcalls context in both.
Do I need to preface the include with a # symbol?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jared Smith
Sent: Monday,
Hi!!
I have a small VPS server in www.eapps.com and im doing some research in
order to install Asterisk in that server..
Does anybody has installed Asterisk in a Virtuozzo VPS System??
Thank you!
Alan
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I'm still struggling to pickup calls. I now have a single context
(entryocginternal) where I have include = parkedcalls.
The log below shows me calling from one internal extension to another, then
picking up, then parking the call.
-- SIP/239-0915d5c8 is ringing
-- SIP/239-0915d5c8
On Mon, Feb 25, 2008 at 3:42 AM, Anthony Messina [EMAIL PROTECTED] wrote:
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
-- Starting simple switch on 'Zap/1-1'
Hi All,
I have installed agx-ast-addons-1.4.5 on Asterisk version 1.4.18. The
problem I have is that RxFAX will not answer an incoming fax. When you
call the number there is just silence. This is over SIP and not ZAP.
The modules rx and tx fax seem to be loaded OK.
core show
I suspect there is something you are not telling us. Try posting this
extension.conf file. Looking at the logs you have here leads me to
believe you have an extension 701 defined to dial SIP/233.
In other words, somewhere in your context is:
exten = 701,1,Dial(SIP/233)
or something very
Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross chan_ss7
but found sifira is not in active development. But is this chan_ss7 stable and
can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7
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