Thank you! I updated the tutorial as well.
l.
2009/5/25 Atis Lezdins
> On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri
> wrote:
> > Hi everyone,
> > after doing the same thing multiple times and struggling to remember how
> it
> > was done, I have prepared a small tutorial that explains how to s
On Mon, May 25, 2009 at 9:52 PM, Olivier wrote:
>
>
> 2009/5/25 eric weaver
>
>> My grateful thanks to whoever can guide me in implementing this...
>>
>> I have a need to place calls via Asterisk Manager Protocol to a legacy PBX
>
> How are both boxes connected ?
>
The Asterisk box speaks via an
2009/5/25 eric weaver
> My grateful thanks to whoever can guide me in implementing this...
>
> I have a need to place calls via Asterisk Manager Protocol to a legacy PBX
How are both boxes connected ?
> and twiddle its MWI lights.
Which manage the phones you're talking about ?
>
>
> By some
original message-
From: "Jimmy Godbout" s...@inbox.com
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
Date: Mon, 25 May 2009 18:01:11 -0800
-
> Check on www.localca
Check on www.localcallingguide.com. You'll find all npanxx that are local to
your exchange.
Jimmy
> -Original Message-
> From: seandar...@gmail.com
> Sent: Mon, 25 May 2009 21:39:30 -0400
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] howto store local exchange pref
I created a mysql table and lookup script for this. One one server were we
could not use mysql, we created an array of exchanges and compared to those.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darc
Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> sean darcy wrote:
>
>> I've looked at the Berkeley DB. That works pretty well, if the exchanges
>> are all stored. But it looks like the exchanges have to be entered 1 by
>> 1 from the CLI. And can only be reviewed, cor
Hi!
> case-2
> Incoming PSTN/ISDN are answered by Fritz, and then forwarded to your
> own Asterisk. Incoming VOIP-calls are answered by your own Asterisk.
- the Fritz!Box usually doesn't "answer" unless you set it up for
voicemail or fax
- define "forwarded": Do you mean "normally" an anlog ph
Nothing is difficult my friend. If you dedicate a few cups of coffee to
it, a couple of days, and do some good googling, you will get it done
yourself.
Good luck!
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torinti
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El domingo 24 de mayo del 2009 a las 19:38:30 -0300,
Daniel Bareiro escribió:
> Now it would remain to find the cause of why I cannot call from a SIP
> extension to an analog telephone. Perhaps it is by something related
> to the contexts in the menti
On Mon, May 25, 2009 at 2:58 PM, John Novack
wrote:
>
>
> sean darcy wrote:
> > The local telco is now going 10 digit dialing even for local (free)
> > calls which used to be 7 digit. For a while no problem, everyone will
> > continue to dial 7 digits, and I'll add the area code. But pretty soon
>
sean darcy wrote:
> The local telco is now going 10 digit dialing even for local (free)
> calls which used to be 7 digit. For a while no problem, everyone will
> continue to dial 7 digits, and I'll add the area code. But pretty soon
> everyone will become used to 10 digits.
>
>
Lucky you.
O
On Mon, 2009-05-25 at 22:19 +0200, Philipp von Klitzing wrote:
> Hi!
>
> > looks interesting, indeed, but as the O.P. wanted to divert PSTN call,
> > one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the
> > hardware of Fritz is capable of it)
>
> "Divert"-ing is a misleading te
If you have asterisk addons installed you can use the mysql
applications to make queries. I find it to be very easy if you know
how to do select and insert queries and understand the basic mechanism
of the dialplan. Other than that, you may want to hire someone to do it.
Sent from my iPod
O
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Hash: SHA1
sean darcy wrote:
> I've looked at the Berkeley DB. That works pretty well, if the exchanges
> are all stored. But it looks like the exchanges have to be entered 1 by
> 1 from the CLI. And can only be reviewed, corrected, or deleted from the
> CLI.
My grateful thanks to whoever can guide me in implementing this...
I have a need to place calls via Asterisk Manager Protocol to a legacy PBX
and twiddle its MWI lights.
By some means I get notified that an MWI light has to be changed.
Via AMP:
1. Send an ORIGINATE command honking up an incoming
The local telco is now going 10 digit dialing even for local (free)
calls which used to be 7 digit. For a while no problem, everyone will
continue to dial 7 digits, and I'll add the area code. But pretty soon
everyone will become used to 10 digits.
There are about 40 3 digit local exchanges. I'
>
> Thanks for your helpful reply.
>
>
>
> I am not so good in coding.
>
>
>
> simply all i need is as follow
>
>
>
> When a call comes, goes into an IVR, and then depending on the entry option
>
> it will connect to a remote SQL Database, to check the pre-existed data,
>
> and in the end of the
Perl and AGI
Piece of cake.!!!
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun"
*---
Hi!
> looks interesting, indeed, but as the O.P. wanted to divert PSTN call,
> one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the
> hardware of Fritz is capable of it)
"Divert"-ing is a misleading term in this case. As I said, use the new
firmware and register Asterisk to th
Un-top-posting...
>> Torintino T schrieb:
>>
>>> Is there any method in Asterisk to enable the updating process into
>>> another SQL database through entering IVR options during the call.
>> Date: Sun, 24 May 2009 22:15:31 +0200
>> From: philipp.kemp...@amooma.de
>>
>> Depending on what you are
Thanks for your helpful reply.
I am not so good in coding.
simply all i need is as follow
When a call comes, goes into an IVR, and then depending on the entry option
it will connect to a remote SQL Database, to check the pre-existed data,
and in the end of the IVR the caller will ent
On Mon, 2009-05-25 at 17:07 +0200, Ngo-Vi Hoai-Anh wrote:
> is installing asterisk directly on FritzBox an option for you? If yes
> I 'v found an interesting link
>
> http://www.ip-phone-forum.de/showthread.php?t=146132
>
>
>
> Manoj Panicker - FOES schrieb:
> > Hi ,
> > Any idea as how
Un-top-posting...
> On Fri, May 22, 2009 at 7:37 PM, Steve Edwards
> wrote:
>
>> On Fri, 22 May 2009, Noel R. Morais wrote:
>>
>>> But I need a way to actively stop it. Without waiting for user hit a
>>> DTMF or the background timeout.
>>
>> What event would trigger your desire to stop the backg
I have a caching name server setup on one of our units but after a prolonged net
outage the internal phones stopped working as well. In searching the bug tracker
I see the bug is still not fixed even though it was thought to be (using
1.6.0.8).
Some suggestions where to set srvlookup=yes but I fa
On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri wrote:
> Hi everyone,
> after doing the same thing multiple times and struggling to remember how it
> was done, I have prepared a small tutorial that explains how to save
> monitored files in different folders per day. This is quite useful
> becauseth
Hi everyone,
after doing the same thing multiple times and struggling to remember how it
was done, I have prepared a small tutorial that explains how to save
monitored files in different folders per day. This is quite useful
becausethe resultingfile system is way more manageable than having maybe
1
I'm planning to play a background music, make some background process and
after that I will play another music or "transfer" the call to another end
point.
I'm gonna see how difficult is to write a function like "StopBackground" to
do that. Any hints?
Thanks in advance,
Noel
On Fri, May 22, 200
is installing asterisk directly on FritzBox an option for you? If yes
I 'v found an interesting link
http://www.ip-phone-forum.de/showthread.php?t=146132
Manoj Panicker - FOES schrieb:
> Hi ,
> Any idea as how to divert the Incoming PSTN calls on the FritzBox
> to one of the Numbers in
I did run make install, probably 3-4 times before I ended up asking that
question in the mailing list.
Here is the required output: to the first one, "could not find module
dahdi".
To the second, it found dahdi in /lib/modules/2.6.18-128.1.10.el5/dahdi
As for the other questions:
What do you do
Hi!
> You mean, you want to use the 7270 as an "isdn-ata" ? perhaps i'm
> wrong, but afaics the pbx-part in any DSL-modem works only on the
> ip-stream (wan/lan).
Both 7270 and 7170 now have a new firmware that comes with a SIP proxy.
It is not fully featured yet (no transfer, NAT/Dyndns issues
On Mon, May 25, 2009 at 08:14:44AM -0400, Mike wrote:
> Hi,
>
>
>
> I've been building a new Asterisk server to replace my previous one, all
> using the latest 1.4.x downloads from asterisk.org.
>
> I can't even get to the point where dahdi works. I have libpri, dahdi and
> dahdi-tools compile
On Mon, May 25, 2009 at 09:32:07AM -0400, Mike wrote:
> Sorry, it seems to have disappeared from my original email!
>
> FATAL: Module Dahdi not found
>
> [snip] all modules listed as not found [/snip]
>
> Error: missing /dev/dahdi!
Your description makes me suspect you have not run 'make instal
Sorry, it seems to have disappeared from my original email!
FATAL: Module Dahdi not found
[snip] all modules listed as not found [/snip]
Error: missing /dev/dahdi!
and /dev/dahdi is indeed absent. How do I make sure it's created as part of
the install process?
Mike
> -Original Message-
On 25 May 2009, at 13:14, Mike wrote:
> Everything compiles seemingly well, but this is what I get when I
> try to start dahdi from the startup scripts (/etc/init.d/dahdi
> restart).
Define 'this'
___
-- Bandwidth and Colocation Provided by http://
He all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
betwee
Hi,
I've been building a new Asterisk server to replace my previous one, all
using the latest 1.4.x downloads from asterisk.org.
I can't even get to the point where dahdi works. I have libpri, dahdi and
dahdi-tools compiled and installed. I have no hardware on that new server,
but I am instal
Hi,
I've been building a new Asterisk server to replace my previous one, all
using the latest 1.4.x downloads from asterisk.org.
I can't even get to the point where dahdi works. I have libpri, dahdi and
dahdi-tools compiled and installed. I have no hardware on that new server,
but I am instal
Hello list,
we have a problem with ISDN-Phones (not cell, not analog) calling into
our Asterisk-Server. Assume we have the number (Germany) +49 123 4567 89
to be called from one ISDN-Phone. I will term (0)123 local dialling
code, 4567 base number of our ISDN-line and 89 the extension (of our
Aster
On Sun, 2009-05-24 at 16:21 +0400, Manoj Panicker - FOES wrote:
> Hi ,
> Any idea as how to divert the Incoming PSTN calls on the FritzBox
> to one of the Numbers in the Asterisk domian? and vice versa.
>
> I want ot use the FritzBox as the bridge between the PSTN and Astrisk
>
> Thanks
> M
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