On 8/14/07, James Collier [EMAIL PROTECTED] wrote:
What if it is an international call? Then your callerID won't work.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: lunes, 13 de agosto de 2007 3:21
Para: Asterisk Users Mailing List -
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
I do something like this. But I am using Realtime and 1.4
context185 1 Set caller_num=${CALLERID(num)}
context185 2 SetMusicOnHold default
context185 3 Playbacksilence2
context185 4 AGI context1.php
AB == Alan Bunch [EMAIL PROTECTED] writes:
AB Just another OpenVPN data point, and not Asterisk related but here
AB goes. I run 15 users over a DSL link on one end and a Internet T1
AB on the other with OpenVPN and it just rocks. The road warrior
AB setup is down to running one script to create
Hi
Does somebody know if I can save the answers made by
the caller person on the IVR menu in a MySQL Database?
If yes, can I save the CallerID as well?
Thanks,
Fabio
Luggage? GPS? Comic books?
Check
, ext-group-home|2000) in new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08:51:35 NOTICE[30952
Anselm Martin Hoffmeister wrote:
I did something similar using multiple records in a row.
Something like
exten = 931,1,Answer()
exten = 931,2,Wait(2)
exten = 931,3,Set(E=1000)
exten = 931,4,Playback(beep)
exten = 931,5,Set(E=$[${E} + 1])
exten =
Hi Fabio,
of course that you can.
One way to do it is working with app MYSQL(), where you will put your sql as
argumment.
read more in http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
good luck,
Thiago Maluf Resende.
2007/8/14, Fabio Ardeola [EMAIL PROTECTED]:
Hi
Does somebody know if
If you want remove in CALLERID.
you can remove it this way:
exten= _X./_+1X.,1, Set()
ok?
good luck!
Thiago Maluf.
2007/8/14, Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten =
On 8/14/07, Thiago Maluf [EMAIL PROTECTED] wrote:
Hi Fabio,
of course that you can.
One way to do it is working with app MYSQL(), where you will put your sql as
argumment.
read more in
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
That's possible, but i wouldn't recommend on large
I have a 536i expansion module attached to a 57i-CT. The BLF lights
on the 536i will light up and work fine for a while... however after a
bit they seem to loose their ability to see if someone is on a phone.
They still work to dial, if I try to dial, however, they don't light
up when someone
On 8/14/07, Atis [EMAIL PROTECTED] wrote:
That's possible, but i wouldn't recommend on large production system.
Using MySQL you would need to connect and disconnect all the time, and
it takes resources.. I would suggest to append that info to CDR
userfield (if you are storing your CDR in
On 8/14/07, Matt [EMAIL PROTECTED] wrote:
I have a 536i expansion module attached to a 57i-CT. The BLF lights
on the 536i will light up and work fine for a while... however after a
bit they seem to loose their ability to see if someone is on a phone.
They still work to dial, if I try to
We have an interesting issue:
One of our providers has two softswitches. Calls coming from the
first one are handled fine by asterisk, calls coming from the second
one and going through the first one are euhm... dropped half a
second into the RTP stream.
I have opened a ticket at Digium for it:
You can eliminate the set CallerID line. This will just set the
variable back to itself. Asterisk will pass the callerid from one
span to the next.
You can use a GotoIF to set the callerid to something else if it is
blank or marked as Private:
exten = s,1,GoToIf($[${CALLERID(num)} = ]?2:3)
There is an example in the asterisk gui (trunk/1.4/asterisknow) that has
cdr-csv parsing.
You could check that, and even use the javascript to generate reports,
integrate it into a little bit of php
and ezPDF generation and bam, you have some reports.
The cdr viewer in the gui is very useful
Good idea! It's working great. I also like your local vs LD logic, much
simpler to do than NXXNXX or 1NXXNXX.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Tuesday, August 14, 2007 8:40 AM
To: Asterisk Users Mailing List -
I just use
exten = +12564286115,1,Goto(${EXTEN:1})
exten = 12564286115,1,noop(It worked.)
I believe that should work
-bk
- Original Message -
From: Thiago Maluf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday,
On Aug 13, 2007, at 4:37 PM, Martin Smith wrote:
See
http://www.asterisk.org/doxygen/1.4/
res__agi_8c.html#c631d48f46d51d4b057
b31807baa1f10
The AGI application will answer the channel if it isn't already
answered.
You probably need to do whatever you want to do in the dialplan, and
James / Atis / Thiago
Let say that the user entry during the call is a
reference number of a house to rent. Would be possible
to check if the reference number is a valid entry on
the MySQL database and then base on its answer define
the next menu item on the IVR menu.
Thanks,
Fabio
--- James
),
priority = mine
ubiphone*CLI
-- Executing Macro(IAX2/ubigradin-2, ext-group-home|2000) in new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw
Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how do I
disable it recognizing #, as it's hanging up my users when they try to enter #.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information
On 8/14/07, Fabio Ardeola [EMAIL PROTECTED] wrote:
Let say that the user entry during the call is a
reference number of a house to rent. Would be possible
to check if the reference number is a valid entry on
the MySQL database and then base on its answer define
the next menu item on the IVR
Shouldn't you ask your attorney these questions?
Any answers you receive here will not legally protect you.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent:
|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08:51:35 NOTICE[30952]: app_dial.c:1076 dial_exec_full: Unable
to
create channel of type 'SIP
Check features.conf
Jeremy Mann wrote:
Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how
do I disable it recognizing #, as it’s hanging up my users when they
try to enter #.
___
--Bandwidth and Colocation Provided by
Ah! you must be American! :-)
Perhaps Zeeshan is looking for an understanding of the issues before
seeking legal advice (it's a lot cheaper that way).
Or perhaps it is a topic worthy of public discussion? I, for one, would
be interested in any known issues.
regards,
Drew
Eric
Does 'sip show subscriptions' indicate that the 57i is still subscribed to
the extension for updates? If not, you might have to do a test with 'sip
Yes it does:
EMSPBX*CLI sip show subscriptions
Peer UserCall ID ExtensionLast state
Type
10.30.17.120 120
Eric Chamberlain wrote:
Shouldn't you ask your attorney these questions?
Any answers you receive here will not legally protect you.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
Nor will any answers from an attorney.
Ever try and get a straight
Is there a way to recognize if someone called our PRI using an 800 number? The
DID is showing my 4 digit primary line, not anything obvious signifying that an
800 number is called?
This e-mail, facsimile, or letter and any files or attachments transmitted with
John Novack [EMAIL PROTECTED] writes:
There is no guarantee of no risk.
Your mother lied to you when she said everything would be alright
Maybe we can convince Digium to have an indemnification program for
people who purchase the business edition! :)
--
Kyle Sexton
Is your dynamic page returning a newline after, like SIP/12345-1\n?
Moj
Mike wrote:
Hi,
Here is my first step (call it a proof of concept) in using the hint
priority with dynamic values.
Background - this works
exten = 12345,hint,SIP/12345-1
To make this a little dynamic, I used
If your 800 number is setup in the same way I understand them to be, the
800 numbers are just forwards to a did (or your main number in the
instance). You'll need to get a specific did setup just for your 800
number to use then you can just recognize the specific DID.
Best Regards,
William J
is there a new way to install asterisk? im using centos 4.5 and trying to
install asterisk. when i do make clean and make install i get this error.
# make clean
--snip--
make[1]: Leaving directory `/usr/src/asterisk-1.2.24/apps'
make: *** codecs: No such file or directory. Stop.
make: ***
Are these POTS lines or a PRI? If you could get RDNIS from the carrier,
then you could tell.
My LD T1 only handles toll free numbers. It is called dedicated as
opposed to switched. I may get some local DIDs from a VoIP provider just
because some providers will reject calls with a toll free
Jeremy Mann [EMAIL PROTECTED] writes:
Is there a way to recognize if someone called our PRI using an 800 number?
The DID is showing my 4 digit primary line,
not anything obvious signifying that an 800 number is called?
Can you just point the 800 number to an unused DID and track the calls
Kyle Sexton wrote:
Jeremy Mann [EMAIL PROTECTED] writes:
Is there a way to recognize if someone called our PRI using an 800 number?
The DID is showing my 4 digit primary line,
not anything obvious signifying that an 800 number is called?
Can you just point the 800 number to an
Hello all,
I've been asked to look into my home dial plan to see if I can improve
it by an important customer (my wife).
What we would like to have happen is that an inbound call rings all the
phones (This is done). Once one phone picks up, of course all the others
stop ringing (Also done).
This legal question pops up every now and then, and depending on how
paranoid you are you can eventually start thinking that the US patent
office is under your bed.(I'm just checking now)
First thing to note is that you aren't worth suing. This is a game that
only applies to very big companies
Here's some details for you all.
Asterisk 1.2
Polycom 301/601 phones
As for my existing dial plan, I'm considering starting from scratch.
Thanks again.
Gerald A wrote:
Hiya,
On 8/14/07, *Russell Handorf* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I've been asked to look
On Aug 14, 2007, at 10:19 AM, Jeremy Mann wrote:
Is there a way to recognize if someone called our PRI using an 800
number? The DID is showing my 4 digit primary line, not anything
obvious signifying that an 800 number is called?
some carriers wont' forward 10 digits DID by default,
On Tue, Aug 14, 2007 at 12:00:25PM -0500, Kyle Sexton wrote:
Can you just point the 800 number to an unused DID and track the calls
by anything coming to that DID? I don't think 800 numbers actually pass
that they are 800 #s.
It has traditionally been the case that non-trunk INWATS rang down
On Tue, 14 Aug 2007, Russell Handorf wrote:
Hello all,
I've been asked to look into my home dial plan to see if I can improve
it by an important customer (my wife).
What we would like to have happen is that an inbound call rings all the
phones (This is done). Once one phone picks up, of
Florent Barbier wrote:
Hi here,
Did you get any solution ? I've quiet the same pb :
http://forums.digium.com/viewtopic.php?t=17394
Thank you for your answer.
flo_turc
Sorry for the late reply :-( We are aware of that particular issue, and
working on tracking it down. Very big sorry
yeah, 'enough' adds back the gray area that the black-and-white 'atomic'
obscures... :P
Moj
Philipp Kempgen wrote:
Mojo with Horan Company, LLC wrote:
set your own mutex using astdb? It may just be atomic enough for you to
get by.
atomic enough - that's a nice term :-)
Gordon Henderson wrote:
On Tue, 14 Aug 2007, Russell Handorf wrote:
Hello all,
I've been asked to look into my home dial plan to see if I can improve
it by an important customer (my wife).
What we would like to have happen is that an inbound call rings all the
phones (This is done).
Well that was it... it is no longer timing out.
On 8/14/07, James FitzGibbon [EMAIL PROTECTED] wrote:
On 8/14/07, Matt [EMAIL PROTECTED] wrote:
I have a 536i expansion module attached to a 57i-CT. The BLF lights
on the 536i will light up and work fine for a while... however after a
bit
After consulting with more experienced folk in the industry, some of which
are running telecom companies for years, I came to the same conclusion what
Henry has said. Now I feel much better and relaxed.
On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote:
This legal question pops up every now
-group-home|2000) in
new
stack
-- Executing ExecIf(IAX2/ubigradin-2,
0|Monitor|wav|20070814-085135-07xx-2000-1187077895.3392.WAV)
in new stack
-- Executing Dial(IAX2/ubigradin-2,
SIP/200SIP/400SIP/600|15|rw) in new stack
ubiphone*CLI
-- Called 200
Aug 14 08
looks broken, is there an apps dir in the source directory?
Mark Quitoriano wrote:
is there a new way to install asterisk? im using centos 4.5 and trying
to install asterisk. when i do make clean and make install i get this
error.
# make clean
--snip--
make[1]: Leaving directory
You want a key system, the fianl frontier of an asterisk implementation,
and currently my holy grail.
The best way to do it in an ugly way is to park the call and have a
speed dial for pickup. Some phones like Aastra 55i and 57i can even have
their hold button reprogrammed to blind transfer to
Erik,
In the sip.conf file, would I put my Asterisk Box's ip address in the
host field? What would I do with the registration field? Leave it alone?
Thanks in advance.
Best Regards,
John
From: Erik Anderson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Digium has a handy tool online!
http://www.digium.com/en/products/voice/audioconverter.php
:-)
--
Chris
Alex Balashov [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On Sun, 12 Aug 2007, MOSBAH ABDELKADER wrote:
Hello all,
have anyone an idea about converting an audio file
Not really sure about SIP exactly, but for Asterisk 1.0 versions, I know
that the Pickup only works with Zaptel channels -- so to use it for any sort
of IP channel, IAX for example, you have to use an addon/patch google
it, 'pickup2' I think it's called works well, allows the Pickup
On Tue, 14 Aug 2007, Anthony Francis wrote:
looks broken, is there an apps dir in the source directory?
Built OK for me:
unicorn*CLI show version
Asterisk 1.2.24 built by root @ unicorn on a i686 running Linux on 2007-08-11
08:22:22 UTC
I didn't do anything special...
Gordon
What did you change?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 14 August 2007 20:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF with Aastra
Well that was it... it is no longer
Anthony Francis wrote:
You want a key system, the fianl frontier of an asterisk implementation,
and currently my holy grail.
The best way to do it in an ugly way is to park the call and have a
speed dial for pickup. Some phones like Aastra 55i and 57i can even have
their hold button
hmmm.. no ideas?! :-|
tom
Thomas Artner wrote:
Hi!
At the moment i am using a digium tdm400 card for my analog phone lines.
The zaptel driver supports fax detection, so incoming faxes are
redirected to the fax extension automatically.
This works without problems with asterisk 1.2.
Since I dont use 1.4 then you tell me. :)
Stephen Bosch wrote:
Anthony Francis wrote:
You want a key system, the fianl frontier of an asterisk implementation,
and currently my holy grail.
The best way to do it in an ugly way is to park the call and have a
speed dial for pickup. Some
Anthony Francis wrote:
Since I dont use 1.4 then you tell me. :)
This functionality is supposed to be supported in 1.4, though I've never
personally tested it. When it's configured it gives the key system
behaviour you describe.
-Stephen-
___
Chris Earle wrote:
Not really sure about SIP exactly, but for Asterisk 1.0 versions, I know
that the Pickup only works with Zaptel channels -- so to use it for any sort
of IP channel, IAX for example, you have to use an addon/patch google
it, 'pickup2' I think it's called works
I need a quick bit of advice from the list.
We purchased an asterisk based phone system back about 6 months ago and
we are using Cisco 7940G phones (I know, not everyone's favorites). We
are using the second line on the phones for paging with a auto-answer,
now my question is having the system
William McCloskey wrote:
I need a quick bit of advice from the list.
We purchased an asterisk based phone system back about 6 months ago and
we are using Cisco 7940G phones (I know, not everyone's favorites). We
are using the second line on the phones for paging with a auto-answer,
now my
Stephen Bosch wrote:
Anthony Francis wrote:
Since I dont use 1.4 then you tell me. :)
This functionality is supposed to be supported in 1.4, though I've never
personally tested it. When it's configured it gives the key system
behaviour you describe.
-Stephen-
On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote:
(the country that still doesn't have universal health care).
Sorry for hijacking this thread but I just couldn't resist.
This is about the only thing in your email I have to disagree with. I
am thankful that we (meaning citizens of the USA)
On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote:
exten=5,2,Dial(SIP/supportSIP/support2,2,tr)
Make this line read:
exten=5,2,Dial(SIP/supportSIP/support2,,tr)
That should do the trick
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
On Tue, Aug 14, 2007 at 06:53:26PM -0400, C F wrote:
Sorry for hijacking this thread but I just couldn't resist.
This is about the only thing in your email I have to disagree with.
Sorry to hijack your thread, but I'll just note that yoou have just
stepped out of this list's topic. So let's
The stability problems we have seem to be related to asterisk crashing
the apache install on the box when the PHP scripts are performing
functions via asterisk. Don't know exactly how they work it all, but
that's the gist of it.
Best Regards,
William J McCloskey
Information Technology Manager
We have differing views. I am a Canadian and was born in the UK
I would not be alive today had it not been for the National Health
Service. I don't see any merit in a system that has over 35 million people
that have no health care and a government that could afford health care
for every man, women
No problem, lets move on..
--
Henry L. Coleman.
Tzafrir Cohen
On Tue, Aug 14, 2007 at 06:53:26PM -0400, C F wrote:
Sorry for hijacking this thread but I just couldn't resist.
This is about the only thing in your email I have to disagree with.
Sorry to hijack your thread, but I'll just
I must agree and I apologize, I agree with Tzafrir.
On 8/14/07, Henry L.Coleman [EMAIL PROTECTED] wrote:
No problem, lets move on..
--
Henry L. Coleman.
Tzafrir Cohen
On Tue, Aug 14, 2007 at 06:53:26PM -0400, C F wrote:
Sorry for hijacking this thread but I just couldn't resist.
On 8/10/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I am having a bit of a problem implementing the pickup command in
my
dial plan. I have setup this rule:
exten = _*8XXX,1,Pickup(${EXTEN:2})
This works as expected when someone dials an extensions number and
I
can get the
As far as I know, yes.
Someone even published a how-to on making up video IVR's.
PaulH
On Sun, 2007-08-12 at 08:54 -0400, SIP wrote:
Is it possible to record or playback a video file in Asterisk?
N.
___
--Bandwidth and Colocation Provided by
You could look at processing the dialstatus (with a goto(s-dialstatus))
as used in macro-voicemail..We did that for a client that got different
beeps, not messages.
PaulH
On Fri, 2007-08-10 at 14:56 +1000, Farooq Ahmed wrote:
Thank you very much who answered to the questions. You have realy
so you are not talking about vanilla asterisk, there are some other
applications involved.
Paging by nature is resource intensive, but still not sure what else is
going on in your system.
On 8/14/07, William McCloskey [EMAIL PROTECTED] wrote:
The stability problems we have seem to be related to
randulo wrote:
Nitesh,
I've messed with the Lumenvox starter kit. If you are serious about
this field, I think it's a must see. It was easy to set up and there
are demos available. Their support is excellent. There is a quiet
mailing list where questions are never ignored and most problems
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