Olivier ha scritto:
...
I didn't get any usable reply yet, beside usual "maybe with next
release".
From http://bugs.digium.com/view.php?id=5014, I don't think "one key
call pickup" is going to appear anytime soon with Asterisk.
Hi Olivier.
That's a pity. ST2030s is in my opinion one of the
Now I have two phones connect to my hardware PBX,and want to Make two calls
from within Asterisk and switch them together. I now have the two numbers
and the other parameter should how to set. for example: the value of data,
type and format ,I set the type "Local" and type AST_FORMAT_SLINEAR
I think you have set the absolute timeout to 3600 sec.
Arjan Kroon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Klaverstyn, David C
Sent: donderdag 21 december 2006 6:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [as
Sounds like a provider or equipment (FXO/FXS call timer) issue. What are
the specs of your system?
Doug
On Thu, 21 Dec 2006, Klaverstyn, David C wrote:
> There seems to be something in Asterisk that disconnects the call at 1
> hour.
>
>
>
> At 59 minutes there is a beep then 1 minute later the c
There seems to be something in Asterisk that disconnects the call at 1
hour.
At 59 minutes there is a beep then 1 minute later the call is dropped.
I have a basic install Asterisk Ver. 1.2.13. I have not specifically
said that calls are to be disconnected at a certain time (not that I
k
I seriously doubt he'd know how to get on the 'Internets'
-Original Message-
From: Doug Crompton [mailto:[EMAIL PROTECTED]
Sent: Wed 12/20/2006 8:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject:RE: [asterisk-users] Re: Match a Numer - the
Maybe if we make enough noise it can be replaced with something better,
rather than just thrown away.
Agentcallabacklogin is great - easy to setup, and provides some useful
functionality.
PaulH
On Wed, 2006-12-20 at 19:09 -0800, Carla Schroder wrote:
> Ick. Don't WANT to do all that. no. A
Your dialplan won¹t hit s-ANSWER Dial doesn¹t return until an answered
call is completed. If you want to announce that the call is recorded to the
person making the call, play it to them before the dial. If you want to
announce it to the person receiving the call, try the M option.
You might a
Sorry, didn¹t realize you were sending the call out on a Zap channel.
Yes, as soon as the call goes out a Zap channel it is ³answered² as far as
Asterisk is concerned. I send out all my findme traffic via SIP.
On 2006-12-19 21:19, "Chris Johnson" <[EMAIL PROTECTED]> wrote:
> On 12/18/06, Eric
Hi Dovid,
> I am actually now working on massproducing door
> openers that will work with asterisk. It will have an
> rj45 port and then a port to plug the door opener in
> to. Please contact me off list if you are interested.
This is an old message, but I was wondering if you are still doing thi
Hello,
I am considering deploying a asterisk system using a VWIC-1MFT-T1
installed in a cisco router.
here is my basic plan:
Telco PRI/T1 <---> cisco 2600 router (with a VWIC-1MFT-T1 card) <--
asterisk server <---> 30 cisco 7960 phones
I have some questions before I spend the $ on this p
How about:
exten => _X.,1,Answer
Does it include all numbers and characters?
On 12/21/06, David Thomas <[EMAIL PROTECTED]> wrote:
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 vis
Ick. Don't WANT to do all that. no. Anyway, according to this it should
still work in 1.4:
http://www.mail-archive.com/asterisk-users%40lists.digium.com/msg166873.html
"Yes, AgentCallbackLogin is deprecated, but it will not be removed until after
1.4. "
On Wednesday 20 December 2006 3:25 pm,
On Wed, 20 Dec 2006, Michael Collins wrote:
> After listing all of that, then give us the description of what needs to
> happen next, the part about deciding which caller ID info to send.
> Pretend like you're explaining it to a bunch of idiots who understand
> only small words and short sentences
Any command to refresh or clear the whole ast database?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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I am having a problem trying to get a Sangoma A101 to work with
Unicall. I have installed the sangoma drivers and everything seems to
be well but when I try to run ztcfg I get the following error:
CAS signalling on span 1 conflicts with HDLC with FCS check on channel
16.
Here is
> 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:
>> http://www.voip-info.org/wiki/index.
On Wed, 2005-08-24 at 12:44 -0400, Asterisk User Group wrote:
> I have three questions about my 7960 phone that I can't discern from the
> docs/wiki.
>
> 1st - If I change the SIPxx.cnf file to change registrations it sets
> up new lines as expected. If I delete a line it doesn't get removed
On Wed, 2006-12-20 at 16:12 -0800, Anthony Rodgers wrote:
> Here's how to unsubscribe:
>
> First, ask your Internet Provider to mail you an Unsubscribing Kit.
> Then follow these directions.
>
> The kit will most likely be the standard no-fault type. Depending on
> requirements, System A and/or S
Jean-Yves Avenard wrote:
What are the advantages of T31 over T30 ?
T.30 is the spec for fax protocol. T.31 is the spec for a Class 1 fax
modem.
So T.31 is a layer that allows something else to perform fax protocol
(T.30) through it.
There may be some intrinsic value to handling faxing
I have used www.ipkall.com I have had one way audio for two weeks now
with no reply from CS.
So I will back you up on this
I guess http://www.kall8.com/ would be the same I think they are one in
the same.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoI
Hi
On 12/21/06, Lee Howard <[EMAIL PROTECTED]> wrote:
spandsp is a dsp library with lots of pieces to it. IAXmodem uses the
T.31 portion (the Class 1 modem) which uses the actual DSP parts of
V.21, V.29, and V.27ter (and potentially V.17). However, the bulk of
the fax protocol is actually per
> -Original Message-
> From: Richard Lyman [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 20, 2006 4:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
> dialplan
>
>
> Douglas Garstang wrote:
Richard Lyman wrote:
Douglas Garstang wrote:
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
*snipped
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies.
When someone calls a numbe
Here's how to unsubscribe:
First, ask your Internet Provider to mail you an Unsubscribing Kit.
Then follow these directions.
The kit will most likely be the standard no-fault type. Depending on
requirements, System A and/or System B can be used. When operating
System A, depress lever and a plast
Is this what you are looking for
exten => _9.,1,Set(CALLERID(num)=3045551212)
exten => _9.,n,Dial(ZAP/g2/${EXTEN:1})
On 12/20/06, Bruce Reeves <[EMAIL PROTECTED]> wrote:
Look at the digit map in your Polycom configuration files. I had the same
problem and had to chage the digit map to supp
You should look at the asterisk-addons package. There is a addon
module in the package called format_mp3 that will play your mp3 files
instead of using mpg123 (which is a dead project).
I just use sox to convert my mp3's to GSM with something like this:
/usr/bin/sox musicfile.mp3 -r 8000 -c1 mu
I am not sure if this is what you are looking for, but I will give it
a shot. There may be a better way to do this but... I would use
agent Queues for your users. Your users can log into the Queue, so
that if the dialed user is not available, then it will drop the caller
into a Queue for a cert
I use Vitality Communications.
http://www.vitelity.net/
I have no problems with the call quality. I have been with them since
they were SixTel.
They have very responsive customer service, a nice provisioning system
and a fairly easy to understand / manage interface.
Cheers, Troy
On 20/12/06,
Douglas Garstang wrote:
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
*snipped
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies. When someone
calls a number in their own company
If you have no statuc stuff in your dialplan, how do you use the 'include =>'
statement? We don't have users... we have companies. It's a hosted IPT
service... and to make the problem even more insane, each company has multiple
levels of organisational structure.
Hardly, you're not required to
You can use Local channels, plus some custom queue_log logging, unless you
want fancy agent/extension association; in that case you'll have to have
your fun with the dialplan :)
l.
On Wed, 20 Dec 2006 23:51:31 +0100, Carla Schroder <[EMAIL PROTECTED]>
wrote:
Sooo what replaces it? Con
Jean-Yves Avenard wrote:
I don't see why rxfax would be less reliable than iaxmodem/hylafax as
it's using the same spandsp to receive fax.
spandsp is a dsp library with lots of pieces to it. IAXmodem uses the
T.31 portion (the Class 1 modem) which uses the actual DSP parts of
V.21, V.29, a
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> > Firstly, in the setup you are envisaging, how do you distinguish which
> > company the caller is calling from? Their extensions number?
> > The context
> > at which th
Hi Phil,
> 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
Sooo what replaces it? Convoluted dialplans? AEL stuff? Agentcallbacklogin was
simple and easy. Hardly anything in Asterisk is simple or easy. Maybe that's
why it was removed. :)
On Wednesday 20 December 2006 2:21 pm, Lenz wrote:
> I would not say that * has been "taken out of the realm of reaso
>I don't see why rxfax would be less reliable than iaxmodem/hylafax as
>it's using the same spandsp to receive fax.
I will defer to Lee Howard on this but IIRC the big factor is ECM which is
not supported in SpanDSP. And another difference is that it is *HylaFAX*
that is recieving the fax itself r
Hi
On 12/21/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
I second that. After struggling with rxfax (which was total cake to set up,
but reception reliability in my specific installation was poor) I bit the
bullet and put in a separate Hylafax server connected to my Asterisk box
with a crossov
> > Firstly, in the setup you are envisaging, how do you distinguish
which
> > company the caller is calling from? Their extensions number?
> > The context
> > at which they enter the dialplan? Or something else?
>
> Good questions, all of them. Unfortnately, I don't have answers to
them. I
> want
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> Ok, but how does that help me? All I want to do is set a variable
DG> to be used later on in the dialplan. Eg, if someone dialls
DG> 2944000, which is in a different company...:
DG> [co1_phone-start]
DG> include => co1_did
DG> include
I would not say that * has been "taken out of the realm of reasonable CC
solutions", luckily there are still a lot of ways to configure * to meet
the most diverse needs, but it's surely a fact that a very convenient and
used approach being deprecated will be an annoyance for CC designers an
On Thu, 2006-12-14 at 13:28 +, Richard Smith wrote:
> Hi all,
>
> I recently installed asterisk 1.2.4 on a HP DL140 G2 server and
> co-located it. My only problem with the box is that there
> is a noticeable delay in the processing of agi scripts compared to any
> other install of asterisk I
On Wed, 2006-12-13 at 12:28 -0500, Matt Van Alst wrote:
> Anyone able to point me the right direction for the following would be
> helpful.
[..]
> Say we have Cisco 7940’s or 7960’s or any phone that has the
> additional buttons other than call appearance. Can we program those
> buttons to start
Hi
On 12/20/06, Lee Howard <[EMAIL PROTECTED]> wrote:
Sure, I guess. The fax detection part comes from Asterisk or OpenPBX or
whatever. Same as with rxfax/txfax, etc.
Well, I know have Hylafax and iaxmodem running on my machine.
Works really well so far and with spandsp 0.0.3
Will see how it
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> > However, don't forget that it searches for matching
> > extensions every time
> > the priority changes. You are not locked into a particular pattern or
> > extension n
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
>> [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(th
> -Original Message-
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 20, 2006 2:41 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: Match a Numer - then continue with,
> dialplan
>
>
> In article
> <[EMAIL PROTECTED]>,
> Douglas Gars
I agree with these fella's, this is a piss poor way of fixing it. I
only know of one call center that used static agents, mostly because
they were sold a peice of crap and they had no idea how to use it the
other way. I think you will find the majority of call centers are
callback centers. This de
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> Benny, lets say I have this...
DG> exten => _X.,1,NoOp(1)
DG> exten => _X.,2,NoOp(2)
DG> exten => _X.,3,NoOp(3) <- Current code execution location
DG> exten => 555,1,NoOp(1)
DG> exten => 555,2,NoOp(2)
DG> exten => 555,3,NoOp(3)
DG>
On Wed, 20 Dec 2006 12:16:45 -0700, Douglas Garstang wrote:
> I think that's the deal breaker right there. I can't start a
> company within an extension. The starting point for each phone
> within a company needs to make extensive use of the include=>
> directive. Features will be disabled by defau
In article <[EMAIL PROTECTED]>,
Tony Mountifield <[EMAIL PROTECTED]> wrote:
> In article <[EMAIL PROTECTED]>,
> Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > > [example]
> > > include => ctx31X
> > > include => ctx3XX
> > >
> > > exten => _X.,1,NoOp(this gets executed first for everything)
> > >
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
> Let's try this a different way. Let's say you have two companies. When
> someone calls a
> number in their own company, we use their INTERNAL caller id. When they call
> someone in
> another company, we want to send
I'm totally at a loss here. I can't get music on hold when placing
someone on hold or when dialing an internal extension. When I dial an
internal extension I hear ringing yet on my phone it shows little
musical notes like it thinks it's hearing music. What to do! :-)
Phil
Lee Jenkins wro
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > [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)
>
> -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 clarifi
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 tha
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,
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 defaul
> -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
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
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 i
> -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]> wr
> -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]> w
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 ex
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
> "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 Goto(glued
> [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
> "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
>
> -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]> wr
"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 mont
> "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".
/Be
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: [asteri
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
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 appre
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 diali
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 gav
> -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 Gars
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 Gars
> -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.
>
>
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, an
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
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 Alperin
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
> -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
> -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 "Man
> -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
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 Co
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 c
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
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 tru
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 Gar
> -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
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 "66
> -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 Se
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 wa
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 --
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 t
>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] w
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:
> http://www.voip-info.org/wiki/index.php?page
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