Hello,
When a user forwards his SIP phone to another extension (say an absent
boss to his secretary) I'd like the unanswsered forwarded call to end up
in the new destination's voicemail. With my current diaplan the call is
handled by the original recipient's voicemail:
Ever since a few weeks ago the connection to FreeWorldDialup stopped
working on our Asterisk server:
This is all we can get out of it:
asterisk*CLI iax2 show registry
Host UsernamePerceived Refresh State
192.246.69.186:4569 814179 Unregistered
Lee wrote:
Maxim Veksler wrote:
I am aware of both of these tools, I don't like them!
They make absolute changes in your /etc/asterisk/* files, they assume
that they are the only thing you will be using for managing your
asterisk pbx and they are both totally unfriendly to 3rd party
changes.
Hi,
sure in an small office you can use iaxmodem/hylafax to receive faxes - we
use it for sending faxes, but would you try to set up about 100 iaxmodems
inside hylafax if you can handle it directly inside asterisk with rx_fax
and a small script ?
[EMAIL PROTECTED] schrieb am 20.12.2006
Olivier ha scritto:
I'm happy to report that with a very litte change to app_devstate.c
(just in the way ast_device_state_changed_literal() is called)
that module just compiles and works fine even without bristuffing
anything.
BTW I'm using a Thomson ST2030S phone with a
Hi
My hardware PBX run asterisk on vxworks,Because the vxworks not support
perl.
Now I want to add a callback function to my pbx.
now it can store Caller and Called party numbers in queue when Called party
is busy
Then I malloc a new ast_channel to call.It is should use
On 12/20/06, Jason Bachman [EMAIL PROTECTED] wrote:
As I understand it, the echo cancelers in Asterisk only work with the
Analog cards (FXS/FXO).
Not true, echo is caused by any number of things in the voice
network, so Asterisk will echo cancel any Zap device. We use it to
cancel ISDN2e and
Astmanproxy is just a proxy. It it just taking the load off asterisk
for multiplexing multiple Asterisk manager connections, but it does
not change the protocol (except to add a couple of features) unless
you select one of the non standard plugins.
Regards,
Steve
On 12/19/06, Daniel Gradecak
Hi Shady,
You'll have better luck posting this to the -biz list. This list is for
non-commercial discussion only.
Alex
On 12/20/06, Shady [EMAIL PROTECTED] wrote:
Looking for a good termination provider for US/Canada
Please contact offlist.
Shady
Hi Timothy,
Mine seems to be working OK as of a few minutes ago:
unlimited*CLI iax2 show registry
Host UsernamePerceived Refresh State
192.246.69.186:4569 727044 216.58.41.183:4569 60 Registered
Do you have any other IAX trunks? Are they
That's odd :)
It's been like this for days I post a message and it's up ? :)
They are now registered :)
Cool.
Alex Robar wrote:
Hi Timothy,
Mine seems to be working OK as of a few minutes ago:
unlimited*CLI iax2 show registry
Host UsernamePerceived Refresh
However
I can call 613 and it works
I can be called and it works
but when I call any other number I get call ended right away :p
Alex Robar wrote:
Hi Timothy,
Mine seems to be working OK as of a few minutes ago:
unlimited*CLI iax2 show registry
Host UsernamePerceived
Below are a few errors in the script and on a google search, although I
found people with the same error, I didn't find a resolution.
Any thoughts on what is causing this error?
Any thoughts as to why the output is not showing on the CLI without doing a
debug?
snip
Content-type: text/html
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG So, in the event that the logic flows beyond
DG coo1_OnNet, we want to reset the caller id of say, 3254001 Doug,
DG to 3254000 Widgets Inc.
DG exten = 3254101,1,Dial(SIP/3254101,20,tr)
DG exten = 3254102,1,Dial(SIP/3254102,20,tr)
DG exten =
Hi List
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Thanks
--
Chris Blunt
Entropy IT Ltd
___
--Bandwidth and Colocation provided by Easynews.com --
You mean that you can't call other FWD users?
Alex
On 12/20/06, Timothy Parez [EMAIL PROTECTED] wrote:
However
I can call 613 and it works
I can be called and it works
but when I call any other number I get call ended right away :p
Alex Robar wrote:
Hi Timothy,
Mine seems to be working
Hello all,
I've seen that the application AgentCallbackLogin()has been set to deprecated
in version 1.4. So I've done some tests based on the tutorial
queues-with-callback-members.txt coming with version 1.4.
What's not clear for me is what is happening to agents.conf, it seems that it's
no
The same with our servers. I just deleted the FWD trunk.
That took less time and quit using the FWD Account
If anyone has any info on why please let me know.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX) 1-563-773-6610 EXT: 250
-- For Information
Hi,
Is AstManProxy an alive project ?
It seems to me that no development are ongoing.
Will AstManProxy comply with Asterisk 1.4 ?
Regards
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
We followed these instructions in trying to eliminate echo:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/doc
s-html/x1695.html
Our lines come in from the telco in a PRI, then connect to a Tadiran
switch which hands the lines off to Asterisk over a T1 card.
I installed the asterisk-addons from source and installed them. It
looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are
compatible with asterisk 1.2.10. Also I unpacked the asterisk source
for the 3 MOH .mp3's and copied them to the appropriate location. Still
MOH is not
Use 'show hints' in the CLI to see if they are actually registering
changing status. It sounds like they're registering but not changing
status.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Tuesday, December 19, 2006 6:49
Indeed,
they can call me,
I can call 613 but not them
Their phone rings for like 1 second.
I get callended.
Alex Robar wrote:
You mean that you can't call other FWD users?
Alex
On 12/20/06, *Timothy Parez* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
However
I can call 613
On Wed, 20 Dec 2006 14:39:42 +0100
Markus Bönke [EMAIL PROTECTED] wrote:
Hello all,
The other thing is, that show agents
doesn't show me which agents are logged in and if I use show queue
I can see local channels attached to a queue but no agents. For my
point of view there is some
Giedrius, did you read my post?
Doesn't seem so as I ask for solution that does NOT require to modify
my dialplan.
On 12/20/06, Giedrius Augys [EMAIL PROTECTED] wrote:
2006/12/20, C F [EMAIL PROTECTED]:
Well I did:
astpp
http://www.astpp.org/
On 12/20/06, Vicky [EMAIL PROTECTED] wrote:
On Wed, Dec 20, 2006 at 02:57:17PM +0100, Olivier wrote:
Hi,
Is AstManProxy an alive project ?
It seems to me that no development are ongoing.
Will AstManProxy comply with Asterisk 1.4 ?
Last release seems to be from 3 monthes ago.
1.4 has not been released yet, as you recall. Anyway,
Olivier ha scritto:
Alberto,
Call pickup is not implemented yet within Thomson ST2030 (1.50 firmware).
More precisely, call pickup current implementation is not Asterisk
compliant.
A new release is scheduled for February (I've got this confirmed by
Thomson 10 minutes ago) but we don't know
Try running it as
./test.php
the hash-bang should take care of the php-location.
The first two lines are one cause of your problem. Could be the lack of
the -q param for php. However, I would expect the script to not show
anything, as it should be reading params from asterisk first. Iirc,
Hi all,
Can someone point me in the right direction here. What I'd like to do
with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom
phones and after the 3rd digit is entered, it dials that extension and
b) dial 9 to get out like older PBX systems. Since my internal
extensions
Look at the digit map in your Polycom configuration files. I had the same
problem and had to chage the digit map to support an extra digit when
dialing 9.
On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote:
Hi all,
Can someone point me in the right direction here. What I'd like to do
with
Florian Overkamp wrote:
Lee wrote:
Maxim Veksler wrote:
I am aware of both of these tools, I don't like them!
They make absolute changes in your /etc/asterisk/* files, they assume
that they are the only thing you will be using for managing your
asterisk pbx and they are both totally
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 10:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On Tue, Dec 19, 2006 at 05:19:57PM
Phil, did you add letter 'm' to your dial options??
exten = _XXX,1,Dial(SIP/XXX,60,m)
Regards
Arlen Nascimento
On 12/20/06, Phil Finkler [EMAIL PROTECTED] wrote:
I installed the asterisk-addons from source and installed them. It looks
like it copied format_mp3.so but I'm not sure if 1.2.14
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG So,
Yes, we have issues with this application being removed as well. In my opinion,
it's a loss of functionality.
-Original Message-
From: Markus Bönke [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 6:40 AM
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
I haven't really been following this thread but doesn't the following
snipet kinda do this
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,playback(international-call)
exten = s,n,playback(please-enter-the)
exten =
Hi Albertore,
As you can guess, my previous reply was mostly based on a general discussion
with Thomson marketing and support teams.
They developped an Asterisk patch to support one key call pickup but never
reached a decision about the way to have this patch maintained as this patch
modifiez
-Original Message-
From: Doug Crompton [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: Match a Numer - then continue with
dialplan
I haven't really been following
Hi -
We'll still need to see more of your dialplan. By your description,
it looks like the call is failing because the Dial() times out.
Take two... My calls are NOT FAILING. Never have so let me restate...
Call comes in receptionist answers. For some ungodly reason this client
does not want
I have 2 sip accounts with name 1234 and abcd respectively. Account
abcd can make call to 1234 but not visa versa. When I change account
abcd to 1abcd, both of them can make call to each others. In the
case, the format of sip account should be start with number. I wonder
whether we can use a
?
KOUCH RACHID a crit:
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject:
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include =
On Tue, Dec 19, 2006 at 10:04:23PM -0500, Jerry wrote:
Heya,
I've got Asterisk 1.2.10 up and running on Debian using the back
ports.
I noticed that it didn't come with mpg123 or depend on it and I
believe
I read somewhere that asterisk now handles it's own mp3 playback? Is
this true?
[EMAIL PROTECTED] wrote:
sure in an small office you can use iaxmodem/hylafax to receive faxes
- we use it for sending faxes, but would you try to set up about 100
iaxmodems inside hylafax if you can handle it directly inside asterisk
with rx_fax and a small script ?
Yes, I would, actually
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
[snip]
[coo1_CallStart]
I'm not sure. I'm a linux newb and this is just running on a server I have
hosted somewhere. I do have control of the box, just not sure what's open
or how to open them.
On 12/13/06, Dovid B [EMAIL PROTECTED] wrote:
You need port 5060 as well as 1-2 UDP open to the server. Also is
-Original Message-
From: Gavin Hamill [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 7:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
On Wed, 20 Dec 2006 14:39:42 +0100
Markus Bönke [EMAIL PROTECTED]
I already posted about this, but contrary to what is stated on the Wiki, mpg123
is required (at least in 1.2.x) if you wish to use mp3's for your MoH.
I decided to go this route:
http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it
--
Kevin Trumbull
-Original
On 12/20/06, Rilawich Ango [EMAIL PROTECTED] wrote:
I have 2 sip accounts with name 1234 and abcd respectively. Account
abcd can make call to 1234 but not visa versa. When I change account
abcd to 1abcd, both of them can make call to each others. In the
case, the format of sip account should
(FYI client did not want VM... Don't ask...)
[general]
static=yes
writeprotect=no
[incoming]
exten = s,1,NoOP(${EXTEN})
exten = s,2,Goto(main-aa,s,1)
exten = 13015550835,1,Goto(main-aa,s,1)
exten = 3015550835,1,Goto(main-aa,s,1)
exten = 5550835,1,Goto(main-aa,s,1)
exten =
www.Kall8.com
Arick Davis
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt
Sent: Wednesday, December 20, 2006 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Need quality toll free 800 number over IAX?
Hi List
I need a quality US
Bzzt. In order to call SetVar, I have to match the extension
dialled. When that happens, there is NO WAY to continue
searching the dialplan after that point for another extension
to match.
You can't use a generic extension and search a database table for
$EXTEN - callerid relation and
On 20/12/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Bzzt. In order to call SetVar, I have to match the extension dialled. When that
happens, there is NO WAY to continue searching the dialplan after that point
for another extension to match.
Can you not use either Goto or the Local
Steve Davies [EMAIL PROTECTED] writes:
Scott Gifford [EMAIL PROTECTED] writes:
[...]
1.5 to 2 seconds. That is a HUGE delay. echo delay is normally
measured in tens or perhaps hundreds of milliseconds, and you are
unlikely to find a software EC that can deal with a 1.5 to 2 second
delay!
On Wed, Dec 20, 2006 at 08:30:27AM -0800, Kevin Trumbull wrote:
I already posted about this, but contrary to what is stated on the Wiki,
mpg123 is required (at least in 1.2.x) if you wish to use mp3's for your MoH.
I decided to go this route:
Does IAXmodem allows you to receive faxes with any extensions
(auto-detecting incoming faxes).
You just let Asterisk do the fax detection for you, and when it hears CNG,
send it to the fax extension, and your fax extension would just Dial() one
of the IAXmodems (using IAX)
[EMAIL PROTECTED]
First off, please, for the love of God, don't cremate me, if I should
already know the answer to this!
I've installed a small setup for an office who wanted to be able to talk to
each other instead of having to rely on MSN to communicate. Weird request, I
know, but hey, we do what we need to do
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Never used them but the rates seems ok : http://www.les.net/
___
--Bandwidth and Colocation provided by Easynews.com --
I have been using an approach such as this but am looking for something
else because of some limitations it has. The phone thinks it dialed,
and was connected to 011 (which it was)
As such, that will be stored in the phones dial history (redial if
nothing else).
I'm not even certain what I
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
Bzzt. In order to call SetVar, I
Douglas Garstang wrote:
Anyone know if there's a way to match a dialplan extension, execute some code,
say set a variable, and then continue with the dialplan?
I want to set a variable when the dialplan flows beyond a certain context. This
would be a great feature.
Match dialed digits of
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 20/12/06, Douglas Garstang [EMAIL
Douglas Garstang wrote:
-Original Message-
From: David Thomas [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 12/19/06, Douglas
I think you're making it far too difficult.
What I do is something like this:
[outgoing]
include = internal
include = longdistance
;Always include internal first, as matches from the first include
;will be used first. This allows you to make sure your internal
;extensions don't go out your
Hi there,
I have a new box setup using the latest version of FreePBX and the
latest SVN of Asterisk 1.2 as of yesterday.
Incoming calls from our PRI work fine. However, outgoing calls gives me
the operator saying The call cannot be completed as dialed after two
rings.
Here's an outgoing
Colin Anderson wrote:
AFAIC, Hylafax + IAXmodem is the way to go for anything serious, unless we
are talking about thousands of users and thousands of faxes per day. I don't
even know what could be scaled to that scenario and not be unmanageable.
For the thousands and thousands scenario you
I don't use many of the features of astmanproxy but it does work. I use
it to capture events from several servers. Some of these are running the
1.4 beta releases.
-Jonahtan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tzafrir
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
Douglas Garstang wrote:
-Original Message-
From: Douglas Garstang
Sent: Wednesday, December 20, 2006 10:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
-Original Message-
From: Eric ManxPower
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
Douglas Garstang wrote:
Change step 2 on your internal extensions to do whatever you want to do
(change the ringer, callID, whatever) then go to main-aa,s,1. Or, change
step 2 to go someplace else, at somplace else, do whatever you want to do,
and then go to main-aa,s,1. The second method is easier to change if, later
I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual
processor machine.
The install lookups on the search for the Sata drive, since however it loads
the sata_sil driver it doesn't work.
Did someone knows what version of Linux is using on Asterisk Now?
Thanks,
Carlos
No, I didn't have m added. Should I have it added? I know I've ran
Asterisk with mp3123 in the past and music worked ok. It seems when I
hit the hold button on the phones, it does trigger the message saying
music on hold is starting but it INSTANTLY stops. I wish it gave some
details as to WHY
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Don't get offended Doug, but I get really frustrated when I try to explain
what I am trying
to do with Asterisk, and people don't seem to quite get it. Your about the
4th person who's
replied to this post, and hasn't
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
I think you're making it far too difficult.
What I do
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 11:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
Used les.net for outgoing for a while, seems to have some bandwidth
problems -- call quality is low.
Time Bandit wrote:
I need a quality US 800 DID over IAX for my Asterisk server,
preferably one
that doesn't cost the earth.
Any suggestions please?
Never used them but the rates seems ok :
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 11:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
Phil Finkler wrote:
No, I didn’t have m added. Should I have it added? I know I’ve ran Asterisk
with mp3123 in the past and music worked ok. It seems when I hit the hold
button on the phones, it does trigger the message saying music on hold is
starting but it INSTANTLY stops. I wish it
Lee Jenkins wrote:
I was wondering the same thing as my MOH isn't working either in a
1.2.14 installation so I'm recompiling mpg123 as per:
http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat
We know you obviously need to use the m flag for the caller to hear
MOH when
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an example of this on the web but I can't seem to find it.
Any advice
Anthony,
Ok I understand. The 011 is unique though and I guess the problem is
the length of the remaining digits. This could vary based on country?? and
I suspect there is no unique rule that could be applied??? I have not
studied this but is there any uniqness to the remaining digits?
Doug
What about comparing the caller id to the dialled number, and if they match,
then call Voicemail() ?
-Original Message-
From: Phil Finkler [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG If I pass a priority, we're right back at square one, we're I'm
DG stuck in a priority and can't get back to an extension.
You ALWAYS have both a priority and an extension. There is no such
thing as being stuck in a priority.
/Benny
www.IPKall.com [EMAIL PROTECTED] wrote:
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Anyone except NuFone.
Their customer service is non-existant - you have to email every day
for a couple of months
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG If I
DG == Douglas Garstang [EMAIL PROTECTED] writes:
-Original Message- From: Benny Amorsen
[mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006
6:16 AM To: asterisk-users@lists.digium.com Subject:
[asterisk-users] Re: Match a Numer - then continue with dialplan
exten =
[example]
include = ctx31X
include = ctx3XX
exten = _X.,1,NoOp(this gets executed first for everything)
exten = _X.,2,NoOp(this gets executed second only if ctx31X
or ctx3XX didnt match)
exten = _X.,3,NoOp(this gets executed third for everything)
[ctx31X]
exten = _31X,2,NoOp(this
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG Surely other people have hit the situation where they first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just
Kevin Walsh wrote:
www.IPKall.com [EMAIL PROTECTED] wrote:
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Anyone except NuFone.
Their customer service is non-existant - you have to email every day
On Wed, Dec 20, 2006 at 02:34:36PM -0500, Phil Finkler wrote:
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
HI
I am able to setup the Dundi and works fine in locating the phone
number's and extensions in branch office's.
Only problem is unable to route the call if we receive it on serverA
from PSTN and some one enter the extension number which reside in
ServerB, it doesn't route the call. But
DG Surely other people have hit the situation where they first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
After doing which, you
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG Surely other people have hit the
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A
users has a
user with extension/callerid XXX, he calls someone in company
B and you
want to set the callerid to company A's main number rather than the
userr's default
Typo, sorry. Should be:
Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 2, Dial(${EXTEN})
;Handle calls from A - B
;Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 1,
I have been speaking privately to a number of CC integrators and resellers
about the AgentCallbackLogin() deprecation issue, and I'd dare say nobody
is enthusiastic about it. With all its problems, AgentCallBackLogin is the
workhorse of most of today's Asterisk CCs, and my impression is
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
Perhaps I can get a clarification
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