hi there,
We just encountered the following.. a customer has a tradifional PBX
that runs next to asterisk. Both PBX's have their own E1 line. Now
'some' numbers are forwarded from the traditional PBX to the new
asterisk server. (both have different DID numbers assigned)
When those numbers are
thanks for your help, I really appreciate itOn 4/25/06, Kevin Smith [EMAIL PROTECTED] wrote:
Yes there is. QUEUE_MEMBER_LIST(queuename)This should return you a list of comman-separated list of the members in
a queue. After that you would need to format it (if needed) so asteriskcan read it back to
Hi list!
I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box.
On the * box I also have a samba share where our CRM app can dump call
files and a cron script is moving the call files every second to the
asterisk directory.
Everything goes really quickly, the call file is placed
Hi,
Check the DISA command.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jnuoiqweahf
kajhdsff
Sent: Thursday, April 27, 2006 12:21 PM
To: asterisk-users@lists.digium.com
Subject:
So sorry, the correct version is 1.2.6 :-)
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Thursday, April 27, 2006 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: Pattern
Hi Chandra, I am also new to Asterisk and I have only just
started installing a test system but I probably can help clarify one or two
things.
I think asterisk "clients" are phones not PCs unless you use"soft
phones" which is software onthe PC(somewhat like Skype) that you
use to
Hi Omar,
Where to dial *+*+#+*+*+# ? If I done it on settings menu, it unlocks the
phone, and than again locks it...
One more question. I have dialplan.xml from 7940 and 7960, can I use it with
7970? I have tried to define it like this
dialTemplatedialplan.xml/dialTemplate
But that doesn't
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
I think is what you are looking for :)
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av jnuoiqweahf
kajhdsff
Sendt: 27. april 2006 06:21
Til: asterisk-users@lists.digium.com
Emne: [Asterisk-Users] treating
Yes, you are correct.I am so sorry. I never use the zap analog card. We only
have one digium T1/E1 PCI card in our small office.
One more question, The analogue zap channel is fxo port? Or fxs port?
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi!
I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time now on my Home PC.
I want to shift to a PC having SATA hard disk .Can I install Redhat 9.0 on SATA hard disk ??some people are telling me that I have to go for Linux Enterprise 4.0.I don`t want to leave Linux 9.0 because I
Rich Adamson wrote:
Mike Fedyk wrote:
Rich Adamson wrote:
Had a Pent 4 server running fc3 crash (kernel panic) and am
I then noticed that FreePBX installed using a SMP kernel (and grub
indicated a non-SMP kernel was installed as well).
Would running an SMP kernel on a Pent 4 potentially
I believe what you refer to is called Ring Back When Free
at least thats how I know it in the UK.
Ah yes, no I remember. We called it Automatic Ring Back.
So we had normal ARB, or ARB on next use.
--
Andreas Sikkema BBned NV
Software EngineerPlaneetbaan
Hi Andrew,
Sorry for my english first.
My configuration and hardware: AAH2.7 2.8, Digium TE100P, welltech 4fxo
voice gateway
SIP Phone
|
|
Asterisk Server - TE100P - Telcom1
|
+ Welltech 4FXO voicegateway Telcom2
Ronald Wiplinger wrote:
I have not used astcc with pin codes so far, since I set-up the phone
number as card number.
Some of my users want now to dial in to the system and than use their
card, which is their phone number.
For that I would need a way of authentication, like a pin.
I want to
Is it possible for asterisk to hang the whole system ??
My Linux box is acting up, and I want to be sure which way to look.
Asterisk or some hardware.
--
Regards,
Nasir.
___
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Asterisk-Users
anyway, you could put the routing stuff in an external file included in extensions.conf:
...
some dialplan stuff
...
[extension-routing]
#include ext-routing.conf
...
some dialplan stuff
...
in ext-routing.conf you have your routing stuff:
exten = 1234567,1,Dial(11)
exten = 7654321,1,Dial(12)
The Hardware support of SATA in RH9.0 is not fully integrated AFAIK , so
moving to a SATA hard disk without an upgrade might not be the safest bet.
on the other hand until you try you won't know for sure .
have you thought of using the Fedora Core ? those have SATA support and
they should be
Hi,
is there a way to completely disable TFTP/HTTP provisioning on the
Grandstream GXP-2000?
Thanks
--
Domenico Viggiani
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
maybe you can try to issue a sip show registry on the console on a regular basis and watch if your * loose registration.
You can also turn on sip debug on the console, to see if the unanswered calls effectively reach asterisk or not. In the latter, is sipphone that loose your registration, so you
Thanx, but for the record and
archive purposes this did not work in 1.2.7.1 but it does work with
1.2.4.
Marnus van Niekerk
tom wrote:
Marnus van Niekerk wrote:
Hi,
I am currently running several * boxes on 1.0.9 with HFC chipset ISDN
modems using i4l's hisax driver and
Hi,
I am running Asterisk 1.2.1 using Digium TDM 400P with 4FXO lines
to connect to the PSTN world. But, I constantly get clipped voice whenever
there is a call placed using Zap channels.
I have tried it all the recommended solutions
- turned off all non essential services on the machine
-
Hi,
Have you try to install this TDM400P card on another asterisk server? Same
problems?
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shyam Gopale
Sent: Thursday, April 27, 2006 5:07 PM
To: asterisk-users@lists.digium.com
Subject:
Hi
I am looking for some advice or tips on how to
make asterisk , to dial a number , when the asterisk
server gets some mail to the asterisk user ,
Is it possible to do so
Guidance requested
Thanks
Joseph John
Hello,
I thought it's exactly what i ask!! Very well!!
Bets regards,
Olivier S.
picciuX a écrit :
anyway, you could put the routing stuff in an external file included
in extensions.conf:
...
some dialplan stuff
...
[extension-routing]
#include ext-routing.conf
...
some dialplan stuff
the most part will be to configure your MTA to trigger a script when the mail gets in. It depends on which MTA you're using.
Once this is ok, you only have, from that script, to generate an auto-dial file to drop in asterisk spool directory to make it dial.
2006/4/27, John Joseph [EMAIL
On Wed, 2006-04-26 at 20:15 -0500, Eric ManxPower Wieling wrote:
Something along the lines of show application retrydial ?
Afaict RetryDial does not allow the caller to hang up the phone and wait
for a call the moment the remote party hangs up. Any way to do this
*without* the caller having to
On Thu, 2006-04-27 at 11:10 +0800, Nathan Alberti wrote:
On 27/04/2006, at 9:15 AM, Eric ManxPower Wieling wrote:
Something along the lines of show application retrydial ?
[EMAIL PROTECTED] wrote:
I am looking for that feature to implement on Asterisk as well.
does anyone know how
Armin Schindler wrote:
On Wed, 26 Apr 2006, Klaus Darilion wrote:
On Sun, April 23, 2006 16:30, Armin Schindler said:
On Sat, 22 Apr 2006, Klaus Darilion wrote:
But I'm still confused. Usually, if a line needs termination, the
termination is needed on both ends. Thus, if there is no line
On 27/04/2006, at 5:45 PM, Patrick wrote:
On Thu, 2006-04-27 at 11:10 +0800, Nathan Alberti wrote:
On 27/04/2006, at 9:15 AM, Eric ManxPower Wieling wrote:
Something along the lines of show application retrydial ?
[EMAIL PROTECTED] wrote:
I am looking for that feature to implement on
kevin ling wrote:
Check the DISA command.
Yup, that does exactly what I need. Thanks!
__
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
On Thursday 27 April 2006 11:08, Ronald Wiplinger wrote:
Ronald Wiplinger wrote:
I have not used astcc with pin codes so far, since I set-up the phone
number as card number.
Some of my users want now to dial in to the system and than use their
card, which is their phone number.
For
On Thu, 27 Apr 2006, Klaus Darilion wrote:
Armin Schindler wrote:
On Wed, 26 Apr 2006, Klaus Darilion wrote:
On Sun, April 23, 2006 16:30, Armin Schindler said:
On Sat, 22 Apr 2006, Klaus Darilion wrote:
But I'm still confused. Usually, if a line needs termination,
the
Unless things have changed, TeleYapper could only accomplish low volume
one at a time calls.
Kerry Garrison wrote:
Asterisk is simply a telephony toolkit, so the simple answer is yes,
Asterisk can do this. Also, being a toolkit means there are a number
of ways to accomplish it. You could
Hello,
I'm writing a small PHP application that generates calls
automatically and tries to store call details on a Mysql
Db, using manager API .
When making an autodial call, I noticed that I couldn't
read $DIALSTATUS values; since I can't evaluate dial
status (BUSY, CONGESTION, NOANSWER),
Finally I'm not sure I found a small compatibility
problem between
chan_misdn and the Romanian implementation of ISDN or
I simply solved a
configuration problem with a huge hammer but I'm
happy it works!
You should try the following combination:
immediate=no
always_immediate=no
---
Amatisoft
--- rommel malana [EMAIL PROTECTED] wrote:
Goodday,
I'm an opensource fanatic and I have already
installed asterisk in my
mandriva linux. Actually, I'm also planning to
install the asterisk
management portal for GUI of asterisk. If anyone
could help me guide
in installing this. Thanks
--- picciuX wrote:
maybe you can try to issue a sip show registry on
the console on a regular
basis and watch if your * loose registration.
Ok:
asterisk1*CLI sip show registry
HostUsername Refresh
State
proxy01.sipphone.com:5060 17476510045105
I was going to buy two units from them. Seeing how
everyone here talks about them I never went thru with
it. Reputation caries a lot of weight.
--- Benchev [EMAIL PROTECTED] wrote:
Thanks Adibar, (sorry List:-)
You have at least an offer. The only thing I've got
so far
was a promiss to
I was going to buy two units from them. Seeing how
everyone here talks about them I never went thru with
it. Reputation caries a lot of weight.
--- Benchev [EMAIL PROTECTED] wrote:
Thanks Adibar, (sorry List:-)
You have at least an offer. The only thing I've got
so far
was a promiss to
Hi,
I'm interested in developing an automated phone
survey and am curious
if Asterisk could be configured to run such a
system.. My idea is to
record a message and a series of sub-questions. The
system would
call each number on a list and play the message,
Depending on the
Is it possible for asterisk to hang the whole system
??
My Linux box is acting up, and I want to be sure
which way to look.
Asterisk or some hardware.
People in the past had the problem. I dont remember
what the cause of the problem was. Try looking at the
archives.
Dovid
hi all,I just installed Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o and have a strange Warning in CLI, which is:WARNING[875]:chan_zap.c:8498 zt_pri_error: 1 TEI remove TEI = 0and another one:
WARNING[875]: chan_zap.c:8498 zt_pri_error: 1 updating callstate, peercallstate 2 to 1Does anybody know what
i am looking for a good ivr system for my company.
these are my question
are there any good ivr's that can be easily
integrated with asterisk ?
and are there any large scale deployment of
asterisk to date ?
Lots of people are using asterisk in a production
enviroment.
When you
On Thursday 27 April 2006 00:25, jnuoiqweahf kajhdsff wrote:
I'm attempting to do this in an AGI program:
I have had *great* difficulty accessing channel variables in *ANY* AGI
language for some time now. I have not filed a bug though, so I am partly to
blame for its not being fixed.
-A.
On Thu, 2006-04-27 at 18:08 +0800, Nathan Alberti wrote:
On 27/04/2006, at 5:45 PM, Patrick wrote:
On Thu, 2006-04-27 at 11:10 +0800, Nathan Alberti wrote:
On 27/04/2006, at 9:15 AM, Eric ManxPower Wieling wrote:
Something along the lines of show application retrydial ?
[EMAIL
JP Carballo wrote:
Yes, certainly, through deadagi.
I just have one question though, why reinvent the
wheel?
There are prepaid systems that work with asterisk.
I have yet to find a prepaid system that allows
multiple concurrent
calls per account. Most seem to be based on a pin
astcc. it comes with asterisk.
--- [EMAIL PROTECTED] wrote:
Any know of any working smart open source billing?
Something smart that can do prepay/postpay and
disconnect customers when they owe or a disconnect a
call in progress for low balance.
___
On Thursday 27 April 2006 03:30, kevin ling wrote:
One more question, The analogue zap channel is fxo port? Or fxs port?
Analog Zap channel or more generally, Analog channel (since chan_modem,
chan_phone, and likely chan_mgcp too) means any channel technology which does
NOT present the
Andreas Sikkema wrote:
I believe what you refer to is called Ring Back When Free
at least thats how I know it in the UK.
Ah yes, no I remember. We called it Automatic Ring Back.
So we had normal ARB, or ARB on next use.
Over the years, traditional pbx manufacturers have implemented
Hi Ronald,
Small mistake, see bellow:
Benchev
Just to give you an idea
I would suggest you to make two .agi files:
astcc.agi and astcc-disa.agi
In astcc.agi you'd leave everithing as it is, and enable
PIN =YES through the astcc-admin.cgi.
Thus you could dial without interogation:
exten =
Tomas Stribrny wrote:
Rich Adamson wrote:
Mike Fedyk wrote:
Rich Adamson wrote:
Had a Pent 4 server running fc3 crash (kernel panic) and am
I then noticed that FreePBX installed using a SMP kernel (and grub
indicated a non-SMP kernel was installed as well).
Would running an SMP kernel on a
Hi,
When I setup a user, I give them an extension like 570xxx. This
is fine and dandy while in one area code, but we've since gone to
other area codes.I'd like the user's to retain the ability to dial
7 digits no matter what number they have. Any thoughts on how to do
that?
EXAMPLE:
A.R. Nasir Qureshi wrote:
Is it possible for asterisk to hang the whole system ??
My Linux box is acting up, and I want to be sure which way to look.
Asterisk or some hardware.
Both are possible. If you watched the cvs/svn commits over the last year
or so, several asterisk issues have
Hi asterisk, openser, ser users.
I have to check video support between asterisk,
open(ser) and rtpproxy .
ASTERISK (b2bua+registrar server)
| |
| |
SER + rtpproxy
| |
NAT
| |
sip agents (with video support)
how much are the codecs thst you cant buy em ? i dont
intend to play judge and jurry however asterisk is a
present that was given to all of us. im some way or
another we should give to those that gave us.
--- Jefferson Carvalho [EMAIL PROTECTED]
wrote:
Thanks for the suggestion ,
But I post
you have all these includes in your (messy) dial plan
yet you didnt post the files that you use in include.
--- Johnny Stork [EMAIL PROTECTED] wrote:
I am new to Asterisk and the protocol/language
complex world of VoIp and PBX. But I have a
dedicated machine running [EMAIL PROTECTED] 2.8, a
Matt wrote:
Hi,
When I setup a user, I give them an extension like 570xxx. This
is fine and dandy while in one area code, but we've since gone to
other area codes.I'd like the user's to retain the ability to dial
7 digits no matter what number they have. Any thoughts on how to do
Hi list!
When I'm doing transfer, to what context/priority does that call goes? Can it
be changed? Is it the same for blind_tr/att_tr/and for transfer that appears
when phone replies with - 302 Moved Temporarily?
The thing is that I'm trying to transfer incoming call from E1 interface back
My Asterisk server believes that a Digium TE405P and sound are
incompatible. Basically, no matter what else I do to the machine it
terms of hardware, if the TE405P is installed, none of the
playback/background/etc commands work. MOH works fine.
So far, I have tried:
1) Seven different PCI
Eric,
Yes.. I am setting calleridnum to be their phone number. And your
example is peachy... except for the fact that it assumes I want to go
out ZAP/g1!!
My problem is I have a very intricite routing plan that routes that
call out several different carriers depending on what you dialed.
(Long
Bob - what type of server/mobo are you using?
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message -
From: Bob McDowell [EMAIL PROTECTED]
To: Asterisk Users Mailing
On Thu, Apr 27, 2006 at 07:39:58AM -0500, Eric ManxPower Wieling scribbled:
Matt wrote:
Hi,
When I setup a user, I give them an extension like 570xxx. This
is fine and dandy while in one area code, but we've since gone to
other area codes.I'd like the user's to retain the ability to
It's a clone built on an Intel 865GBF.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
Andrews
Sent: Thursday, April 27, 2006 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE405P vs.
Hello folks!
I'm trying yo set up a email2fax and fax2email on my asterisk box.
The rxfax works fine in my setup.
The problem is with the txfax.
I have tryed all snadsp version (0.0.2x and 0.0.3x) but I get this
errors. Because I can't find anything on Internet I'm hoping u can give
me a hand.
HI
I am trying to establish a connection between ASTERISK and ALEPO but I can
not,
since you have reached to make them communicate can you help me with the
changes made to asterisk, in this way I will be able to check if the
problem is the same with my ALEPO .
I would appreciate every help
Ok that works... and I could do that if all I cared about was added
the 1 or country code. I guess in theory I could set a variable
set(ARECODE=${callerid(num)0:3}
_ = do stuff here for 7 digits
And then transform the number by taking the areacode and putting it in
front of the
[from-pstn]
include = from-pstn-custom ; create this context in extensions_custom.conf
to include customizations
include = ext-did
;exten = fax,1,Goto(ext-fax,in_fax,1)
exten = _.,1,Wait(1)
exten = _.,2,Goto(from-pstn,s,1)
Here is what is happening :
Your ZAP channels are in the context
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
Matt wrote:
Eric,
Yes.. I am setting calleridnum to be their phone number. And your
example is peachy... except for the fact that it assumes I want to go
out ZAP/g1!!
My problem is I have a very intricite routing plan that routes that
At $10.00US per concurrent channel, it is better to buy, than to
complain. Do you complain i someone gives you a new car but you have to
pay for the gas?? (Bad example with Oil prices going high, but you get
the point)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911. Is there a way to use asterisk to seize a phone line from the fax machine?
I don't want to have to have an analog line that only
That will work? So if I have:
CALLERIDNUM = 5705551212
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
exten = _570NXX,1,Dial(Zap/g1/${EXTEN},1)
And if CALLERIDNUM = 7175551212
exten = _717NXX,1,Dial(Zap/g2/${EXTEN},1)
(Notice 717 calls go out g2.. and 570 go out g1).
That
On Thursday 27 April 2006 07:52, Rich Adamson wrote:
Not likely either form can truly be implemented in a agi script without
significantly impacting other pbx functions.
I dunno... off the top of my head:
- improve upon the standard extension macro such that any Dial() uses 'g'
option, and
Hi,
I am getting this message on the * console on my first pri span. Pri
show span show it is down, and I can't make any calls from the span.
Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Apr 27 07:40:23
Joe Pukepail wrote:
I have a question, we have some locations were I'm just planning on
putting in a PRI, management also wants analog lines incase the PRI is
down and someone calls 911. Is there a way to use asterisk to seize a
phone line from the fax machine?
Multiple ways to do that.
Hello,
Is there Somebody to provide me a DID numder on a voip
gateway which one support t.38 to test FOIP ?
Regards
Harry
___
Faites de Yahoo! votre page d'accueil sur le web pour
Dearest List,
I understand how to handle guest calls via SIP and IAX. However, when such
a call is placed, it will not look like IAX/guest-1234, or SIP/guest-1234.
Instead, it will be something like IAX/the.callers.ip.address-1234
My issue is with getting this to map to a Flash Operator Panel
Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not
seem to be a menu area/settings for Incoming Calls?
If you have a similiar setup, or know what the settings should be, could you
possibly post them? If I were to create a dial group
to ring all extensions, could that
For instance, I have tried the 2 below, but still it does not ring an existing
extension, although the logs show it trying
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
;exten = fax,1,Goto(ext-fax,in_fax,1)
Johnny,You need to setup an Inbound Route that matches all DIDs and all CIDs. In FreePBX, click on Inbound Routes, create a new route with blank CID and DID, and point it where you want it to go. It should work after that.
AlexOn 4/27/06, Johnny Stork [EMAIL PROTECTED] wrote:
Since I am using
On a related issue, at locations where we have 3 or 4 phone lines connected
to asterisk and they are all in use and someone dials 911 we want it to
disconnect one of the active calls so the 911 call can be made. Does
anyone know how to do this? Would I need to use a device like the above or
Quoting http://www.asteriskguru.com/tutorials/e1t1.html
-- configuration on SBC.
If you are being flooded (several times a second, non stop and the pri
never worked) by lines as:
Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
;exten = fax,1,Goto(ext-fax,in_fax,1)
exten = _.,1,Wait(1)
exten = _.,2,Goto(from-pstn,100,1)
Try somethin like
[from-pstn]
include = from-pstn-custom ; create
Also
If you see the error Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874
pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span
1
only occasionally, then you might have some devices in your pc (ide
cards?) taking to long when taking an intterupt.
You might want to try to put
Hi, we've had a couple of Sounstation 4000's around for a couple of
months working fine with our * box.
Today, I tried for the first time to do a local 3-way conference with
one of them, and could not find the confrnc soft key for doing that
(as stated in the user manual). Spent 20 minutes without
Remco Barende wrote:
Hi list!
I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box.
On the * box I also have a samba share where our CRM app can dump call
files and a cron script is moving the call files every second to the
asterisk directory.
Everything goes really quickly,
Thnks for the link. However, I know span 1 is pri because when I add the
vpmsupport=0 parameter when loading wct4xxp, everything works and those
messages don't show up. I think vpmsupport=0 parameter disable echo
cancellation on the board (I have TE411P card).
-Original Message-
From:
Ok here is what we did.. all in one context:
exten = _NXX,1,NoOp(Customer Area Code Is: ${CALLERIDNUM:0:3})
exten = _NXX,2,Goto(${CALLERIDNUM:0:3}${EXTEN},1)
exten = _1NX,1,NoOp(Chopping One Off Number: ${EXTEN:1:10})
exten = _1NX,2,Goto(${EXTEN:1:10},1)
Had to fix Eric's
I have a single scsi drive in the system. In a week or so, we will
replace it with a sandisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
Blades
Sent: Thursday, April 27, 2006 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I write perl applications for a living and have developed code to talk to all kinds of hardware. What I'd like to do is pull a list of phone numbers from sql via dbi and call each. An initial voice messsage would be played asking the recipient if they'd optionally like to fill out our survey. If
Matt wrote:
That will work? So if I have:
CALLERIDNUM = 5705551212
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
exten = _570NXX,1,Dial(Zap/g1/${EXTEN},1)
And if CALLERIDNUM = 7175551212
exten = _717NXX,1,Dial(Zap/g2/${EXTEN},1)
(Notice 717 calls go out g2.. and 570 go out
On 4/27/06, Rich Adamson [EMAIL PROTECTED] wrote:
Joe Pukepail wrote: I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is
down and someone calls 911.Is there a way to use asterisk to seize a phone line from
I have a Dell PE SC420 (a no-no with a TE110P) connected to a Mitel SC-200. The Mitel gets Slip and Frame errors that cause the T1 card in the Mitel to go offline and this causes a service interruption. Could the SC-420/TE110P be causing these errors? I know it is listed on the incompatibility
Hi Armin!
Armin Schindler wrote:
I'm not aware of such a cable to buy. Normaly, when you create a NT-side the
connection is not made with just one cable (like I did because both device
are just 10cm away from each other). In most cases you have an ISDN bus
cabled in the rooms where the
Joe Pukepail wrote:
I have a question, we have some locations were I'm just planning on
putting in a PRI, management also wants analog lines incase the PRI is
down and someone calls 911. Is there a way to use asterisk to seize a
phone line from the fax machine?
I don't want to have to
I actually tried
that before but it didnt seem to work. I tried once again and still nothing
rings, whether I set the destination to a single extension, or a ring group. But
the suggestion from another user below did work, but wont go to voicemail yet
when its not answered.
[from-pstn]
Hi Eric,
this is my zapata.conf (zap/1 is a FXS but not used during tests):
;-
; Channel: zap/2 [in] - Telecom (lasciare libera)
;-
language = us
musiconhold = default
signalling = fxs_ks
channel = 2
Could you kindly let us know what numbers those survey-calls will be
coming from, so we can all add them to our blacklists? Thanks!
TV JOE wrote:
I write perl applications for a living and have developed code to
talk to all kinds of hardware. What I'd like to do is pull a list of
phone
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Hash: SHA1
Hi,
Is it possible to get the callergroup or pickupgroup of a phone in the
dialplan? So I can make decisions depending on the caller/pickupgroup.
chris...
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Comment: Using GnuPG with
On Thu, 27 Apr 2006, Klaus Darilion wrote:
Hi Armin!
Armin Schindler wrote:
I'm not aware of such a cable to buy. Normaly, when you create a NT-side
the connection is not made with just one cable (like I did because both
device are just 10cm away from each other). In most cases you have
I actually tried that before but it didnt seem to work. I tried once again
and still nothing rings, whether I set the destination to a single
extension, or a ring group. But the suggestion from another user below did
work, but wont go to voicemail yet when its not answered.
[from-pstn]
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