dave cantera wrote:
remco, et al,
could I use dundi where I could use an area code to determine the
connecting server or dial string? just like we would use 88XXX to dial
a 3 digit extension on another server at location 88? or dial 84XXX for
a 3 digit extension on a server located at
lenz wrote:
Is the queue enidan configured at all in queues.conf? and how is it
defined?
l.
Sorry, I should have added that:
from queues.conf:
[enidan]
strategy = ringall
;announce = enidan-queue
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL
On Tue, 15 May 2007, Vincent Delporte wrote:
Hello,
In case there are other users of the AsteriskWin32 port...
I haven't really used the AGI feature of Asterisk to run an application
from extensions.conf. *Win32 supports Perl, which I don't know. Apparently,
it's also possible to
Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c , And I already tried
with 'spandsp' application for this. But I am getting errors.[I followed the
instructions at http://www.soft-switch.org/installing-spandsp.html]
I'm using an FastAGI written in Delphi for my IVR so I can confirm it
works just fine. I wrote all the code from scratch and it wasn't a big
deal, but you can find sample code on Free Pascal sites (google will
help you).
Also I'd recommend turning your idea into an FastAGI. It will work with
both
Hello,
Is your userfield type varchar(255)?
Also check if you edited the cdr_addon_mysql.c and Make file to tell
cdr_addon_mysql.c to store uniqueid as outlined here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062)
On 5/10/07, Steve Kennedy [EMAIL PROTECTED] wrote:
On Thu, May 10, 2007 at 06:36:53PM +0200, nik600 wrote:
i have a Te205P connected to a PRI E1, can i force the outgoing
callerid to change for each context?
for example:
[outgoing_context_one]
;force callerid to 12345
exten =
Christopher Robinson wrote:
When I make a regular call from my SIP phone connected to my Asterisk
server I have no issues, however when I make a call using Originate :
'Channel'=SIP/[EMAIL PROTECTED],
'Context'='mycontext',
'Exten'='899',
'Priority'=1,
'Callerid'='whatever'));
It creates a
Do any one knows the formula to calculate memory and cpu usuage for channel
on g729 codec,to know the hardware required for 100 concurrent call.
Regards
*
No employee or agent is authorized to conclude any binding agreement on
Does such formula exist ?
And do you have other functions/apps that demands cpu power that needs to be
taken into the formula.
And please skip that disclaimer you have in the bottom !
/Mats
On 5/15/07, Khaled Chehab [EMAIL PROTECTED] wrote:
Do any one knows the formula to calculate
If you have the clipping issue, make sure you get HPEC version 8.2
from Digium.
Note, however, that we have observed stability issues with HPEC 8.2 (two
kernel panics in two weeks since installing HPEC). (The 9.00 version had
such severe clipping that we could not run it long
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
All this seems to be working. However, we just don't feel we are doing
things properly and reading up on the wiki more about AGI and dialing
out, etc, just makes me feel we could be doing things better.
Here are some of the things we
On Sun, May 13, 2007 at 01:00:27PM +0100, --[ UxBoD ]-- wrote:
Hi,
I have been using Zapateller with a TDM400 no problems at all, but
recently I have ported our BT number to a VoIP provider, and have a
strange problem. When I phone our number I first get the BT
unavailable three tone
Hmmm, I do have answer() in the dialplan. When I was using with the BRI I did
get the single SIT tone. My intention for Zapateller is to play the SIT tone
to try and stop autodiallers.
On Tue, 15 May 2007 12:00:35 +0100, Phil Reynolds [EMAIL PROTECTED] wrote:
On Sun, May 13, 2007 at
On Tue, May 15, 2007 at 12:23:56PM +0100, --[ UxBoD ]-- wrote:
Hmmm, I do have answer() in the dialplan. When I was using with the BRI I did
get the single SIT tone. My intention for Zapateller is to play the SIT tone
to try and stop autodiallers.
Single SIT tone? Do you mean you now get it
Hi,
I'm tryng to place a call through Asterisk Manager Originate Action.
Since I want separate CDR for each of the two legs of the call, I'm
forking CDR with ForkCDR as the first Channel has picked up.
The problem is that, while the first CDR is fine, in the second one the
answer field is
Could I rewrite this in Delphi instead?
I never used Delphi to write an AGI but I've seen a class in
FreePascal that you could probably use as a base :
http://www.automated.it/asterisk/fpc-agi.html
hth
___
--Bandwidth and Colocation provided by
Hi,
'sip show peers' display info this way
Name/username HostDyn Nat ACL Port Status
229/229xxx.39.12.58 D N 63969OK (26 ms)
228/228xxx.39.12.58 D N 63961OK (32 ms)
But this makes is difficult
On 5/15/07, lizhong zhu [EMAIL PROTECTED] wrote:
I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run
./configure and menuselect with embedded modules. but running make comes out
errors:
ranlib libmxml.a
[...[
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:23:
It may not be exactly what you are looking for, but agentcallbacklogin
with ackcall=yes requires the # key to be pressed to answer a call from
a queue. We use this to avoid the possibility that the call ends up in a
cellular or home voicemail.
You can set the queued call to ring instead of
What about using a Lucent TNT and an asterisk box. Is the limit the
TDM--G.711 conversion?
On 5/14/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Mon, 14 May 2007, Kapil Dhawan said something to this effect:
I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is
it
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk
Servers http://www.voip-magazine.com/content/view/3644/0/1/0/ in the
VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
On May 14, 2007, at 11:27 PM, Atlanticnynex wrote:
I'm curious what kind of configuration/features/modules you could
recommend for my setup. Can you explain further what you mean by
OpenSER to Asterisk?
If you want to go Open Source, I think OpenSER is a good choice. You
won't need to
Hello everyone,
I am new to Asterisk and I am trying to find the list of sip soft phones list
but I am having trouble finding the list. Can some one point me to a url where
I could find this? I have tried looking for this myself and found it twice but
now I can't find it again. Thanks much.
I recently got a polycom 501.
I was trying to get the phone to accept the TFTP boot files.
I was REALLY confused when I finally figured out that
the phone does FTP by default and you have to go change it to TFTP using the
keyboard menus to switch it to TFTP.
Am I missing something here? I
On 5/15/07, Scott Berry [EMAIL PROTECTED] wrote:
I am new to Asterisk and I am trying to find the list of sip soft phones
list but I am having trouble finding the list. Can some one point me to a
url where I could find this? I have tried looking for this myself and found
it twice but now I
On 5/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk
Servers in the VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten =
To answer my own message, I figured out a solution (untested) about 10
minutes after posting and leaving the office. Doh!
Anyway, the solution (now tested) was to make the Asterisk server behind the
NAT register with its peers. Despite reserving port 4569 in the firewall,
that was not enough in
Hi:
Jerry Geis wrote:
I recently got a polycom 501.
I was trying to get the phone to accept the TFTP boot files.
I was REALLY confused when I finally figured out that
the phone does FTP by default and you have to go change it to TFTP using
the
keyboard menus to switch it to TFTP.
You
Kristian Kielhofner wrote:
On 5/15/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of
Asterisk Servers in the VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten =
Hello,
I'm writing because we have problems with an asterisk installation
(Trixbox ver. 1.2.3). We have a customer which is receiving a lot o
telephony traffic (more or less 1 call/2 min.); we are using a TDM400
board, with 3 PSTN lines configured and we have two big issues:
- Calls are
Hello,
I guess you have something like this:
ACTION: Originate
Channel: Local/1234
Exten: 4321
Priority: 1
Context: blabla
And in
[blabla]
Exten = 4321,1,Dial(something.
Instead use magic /n setting with Local channel. See here:
http://www.voip-info.org/wiki/view/Asterisk+local+channels
Perfect, Auriol!
Exactly this.
Cléviton.
-Mensagem original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Seb Auriol
Enviada em: terça-feira, 15 de maio de 2007 12:45
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Assunto: RE: [asterisk-users] Mr.
On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote:
Hello,
I’m writing because we have problems with an asterisk installation
(Trixbox ver. 1.2.3). We have a customer which is receiving a lot o
telephony traffic (more or less 1 call/2 min.); we are using a TDM400
board, with 3 PSTN
Two problems, possibly related:
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
On Tuesday 15 May 2007 19:11, Dave Cotton wrote:
On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote:
Hello,
I’m writing because we have problems with an asterisk installation
(Trixbox ver. 1.2.3). We have a customer which is receiving a lot o
telephony traffic (more or less 1 call/2
It has nothing to do with the GUI. Trixbox compiles Zaptel for you and
provides them as RPMs for installation. Removing the RPMs and all the
configs they leave lying around and compiling from source can be a
complicated process, and the Trixbox forums/mailing lists will be better
able to help the
On Tue, 2007-05-15 at 19:57 +0300, Diego Iastrubni wrote:
On Tuesday 15 May 2007 19:11, Dave Cotton wrote:
Contact the Trixbox mailing lists?
Why is that? You think some fancy-shmancy GUI will fix this? The problem is
obviously in the zaptel area. But hey... this is asterisk-users...
/me
On 5/15/07, Seb Auriol [EMAIL PROTECTED] wrote:
Kristian, I think Cleviton's point was that [lookupdundi] does not appear to
be a macro, and so how can it have arguments if it just a normal context /
extension?
Kind regards,
Sebastian
Sebastian,
Reading the post again I could see it
Per Jessen wrote:
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk -
at which time I've more than once seen the
On 5/15/07, J. David Bavousett [EMAIL PROTECTED] wrote:
Two problems, possibly related:
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
David:
Thanks for your tips...all but one issue, then, solved as best it can
be solved. Just the outbound dialing issue, now...
--David B.
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--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
On Tue, 2007-05-15 at 19:16 +0200, Dave Cotton wrote:
Perhaps the fancy-shmancy GUI is hiding the configs.
Al Bochter has just told me off list that Trixbox is Asterisk
But according to their site
trixbox is a complete application platform. When you install trixbox you
have a powerful
Just to be clear:
1) I do not use to configure the * config files with trixbox GUI but I manually
edit the file
2) from my point of view, the main advantage of trixbox is to have an *
installation uprunning in half an hour, then it's up to you use the GUI or
manually edit files
3) I did not
If you use edit the config files on a trixbox system like you would on
an * box, any time you reboot or hit the red update bar, it will reset
the files to what the gui has. The only files you can edit on a trixbox
system are the _custom.conf files. This may be the issue with the time out
Martin
On 5/15/07, Marco Vescovi [EMAIL PROTECTED] wrote:
Just to be clear:
1) I do not use to configure the * config files with trixbox GUI but I
manually edit the file
2) from my point of view, the main advantage of trixbox is to have an *
installation uprunning in half an hour, then it's up to you
Try the 'g' option to VoiceMail().
CP
Stephen Bosch wrote:
Hi:
I have a user saying that the volume of voice mails is too low.
Is there a way to tweak the recording level for voice mail?
-Stephen-
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Dave
Please note what the core is..
* Asterisk(tm) Open Source PBX
The GUI only writes some of the conf file for you.
So if there is a fix for the list member that works on Asterisk please help
them out.
I have worked on other Asterisk based PBX systems and the conf files are just
Hi,
I have purchased a TE212P (Dual Span) Digium card and have compiled
Zaptel 1.4.2.1 and LibPri 1.4.0 and Asterisk 1.4.4
ztcfg -vv shows this:
Zaptel Version: 1.4.2.1
Echo Canceller: MG2
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN
On Tue, 15 May 2007, Per Jessen wrote:
Per Jessen wrote:
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk -
at
On May 14, 2007, at 1:55 PM, Daryl Jurbala wrote:
On May 14, 2007, at 1:29 PM, Zoa wrote:
Several people do use it for handling 50k minutes a day. (I'm one
of them).
Yes, you need to know what you are doing, and have a nice design, but
it is possible.Our code is only slightly altered.
Hiya everyone. I have been working on a fun little app to watch what's
going on in your asterisk box via its manager interface. There's a
screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it
requires allegro, but I was more keen about getting the ideas down than
worrying
Are you sure that you set the T1/E1 jumpers on the board correctly for
E1 mode?
Matthew Fredrickson
On May 15, 2007, at 1:23 PM, Matt Brown wrote:
Hi,
I have purchased a TE212P (Dual Span) Digium card and have compiled
Zaptel 1.4.2.1 and LibPri 1.4.0 and Asterisk 1.4.4
ztcfg -vv shows
On May 15, 2007, at 1:50 PM, Khaled Chehab wrote:
Do any one knows the formula to calculate memory and cpu usuage for
channel on g729 codec,to know the hardware required for 100 concurrent
call.
I believe you can for sure do nearly 100 calls (on pretty much any
decent system) with
I must clarify, when I experimented with gastman, it was on the Windows
platform only. Thus, I can understand why it didn't seem to be updated
very often. Now that I have entirely switched over to *nix, I have NOT
experimented with gastman. I'm sure it has had the opportunity to far
surpass
I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way
to integrate them.
Two questions arise:
1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI
system? The idea is to intercept outbound calls from the Nortel PBX and
redirect them via VoIP
Gordon Henderson wrote:
You're getting the error message because ${CALLERID(num)} is empty.
ie. there is no caller-Id set, so I'd work on working out why there's
no callerId set for the very first call...
Eg. start with:
exten = _X.,1,Noop(CallerId is ${CallerId(all)})
exten =
Vincent Delporte wrote:
Hello,
In case there are other users of the AsteriskWin32 port...
I haven't really used the AGI feature of Asterisk to run an application
from extensions.conf. *Win32 supports Perl, which I don't know.
Apparently, it's also possible to write AGI applications as EXE's
On Tuesday 15 May 2007 3:31 pm, Jeremy Mann wrote:
1.Is it feasible to use asterisk as a Man in the Middle for a T1
PRI system? The idea is to intercept outbound calls from the Nortel PBX
and redirect them via VoIP to another asterisk box at another branch
transparently(thus saving
Vincent Delporte wrote:
Hello,
In case there are other users of the AsteriskWin32 port...
I haven't really used the AGI feature of Asterisk to run an application
from extensions.conf. *Win32 supports Perl, which I don't know.
Apparently, it's also possible to write AGI applications as EXE's
On 5/15/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a ton of Nortel MICS/CICS phone systems and am looking for an easy
way to integrate them.
Two questions arise:
1.Is it feasible to use asterisk as a Man in the Middle for a T1
PRI system? The idea is to intercept outbound
On Fri, May 11, 2007 08:21, Gordon Henderson wrote:
On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote:
If you think your ISP is reliable enough then go for it!
I've had less ADSL issues last year than ISDN issues! ;-)
(And that while ADSL is running over that very ISDN line!)
On Fri, May 11, 2007 10:31, Chris Bagnall wrote:
There is a small (and growing!) number of small businesses (and not so
small ones either!) who are moving towards using their broadband
(typically ADSL in the UK) connection for Telephony - and even
installing
a 2nd ADSL line just for VoIP.
Never received a response for this from anyone. This is being seen more
frequently now.
Please Suggest.
Regards,
Sanjay Rajdev
- Original Message -
From: Sanjay Rajdev [EMAIL PROTECTED]
To: asterisk-users asterisk-users@lists.digium.com
Cc: asterisk-dev [EMAIL PROTECTED]
Sent: Friday,
The only thing I'd probably lose is the ability to do faxes! So I am going
to investigate that further first!
Havn't doen that in years - an online fax service sends me my faxes by
email and I sent out faxes through them, not that I ever do that.
--
Chris Mason
(264) 497-5670 Fax: (264)
Per Jessen wrote:
Gordon Henderson wrote:
You're getting the error message because ${CALLERID(num)} is empty.
ie. there is no caller-Id set, so I'd work on working out why there's
no callerId set for the very first call...
Eg. start with:
exten = _X.,1,Noop(CallerId is ${CallerId(all)})
Vincent Delporte wrote:
Hello,
I'm trying to fill CID data into the astdb using AsteriskWin32's
asterisk.exe, to no avail: The batch file stops after the first line,
and just waits:
rem c:\cygroot\mystuffimport.bat
rem
rem
Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk
Servers http://www.voip-magazine.com/content/view/3644/0/1/0/ in the
VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
Sanjay Rajdev wrote:
Never received a response for this from anyone. This is being seen more
frequently now.
Please Suggest.
Regards,
Sanjay Rajdev
- Original Message -
From: Sanjay Rajdev [EMAIL PROTECTED]
To: asterisk-users asterisk-users@lists.digium.com
Cc: asterisk-dev [EMAIL
I use asterisk -rx database put value if you are trying to batch it
from windows you can use plink
On 5/15/07, Vincent Delporte [EMAIL PROTECTED] wrote:
Hello,
I'm trying to fill CID data into the astdb using AsteriskWin32's
asterisk.exe, to no avail: The batch file stops after the
There's a better way. Take a look at how to do a Find me at
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
You can have the call only completed when they press a key on the receiving
phone. No voicemail will trigger that.
Chip Schweiss
-Original Message-
From: [EMAIL
Jeremy,
Both 1 and 2 are feasible and have been done by many people including
the company I currently work for and the company I previously worked for.
For the analog lines, I would recommend a channel bank with analog
ports. If you want to redirect inbound calls on an analog line as well
I have built Asterisk 1.4.4 on my Solaris 10 x86 box:
LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib'
CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw
--without-oss --without-vpb --prefix=/opt/asterisk-1.4
The build and install go fine but the asterisk
Hello
I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I
also have an IP phone in a remote network across the Net. The PBX +
gateway, and the phone are both behind a NAT router.
I was wondering:
1. When a customer calls us through the POTS line and I pick up the
hello, James FitzGibbon:
thank you for your help. i am very new to arm-linux and embedded linux. i think
what you said is right. i am not very sure the steps i taken are correct. i
post it here and please give me some help. it might be help other arm-linux
users too. i installed all necessary
Hi Everyone,
I have an asterisk box in my office. It does not display the correct Incomming
Caller id.
For incomming we are using ISDN Bri line which is terminated in a Digium 4 port
bri card (B410P).
Like if a number say 02 12345678 calls to our line asterisk displays it 12
12345678.
I forgot to give the asterisk logs
pbx*CLI
-- Executing Set(mISDN/2-2, FROM_DID=3722) in new stack
-- Executing Gosub(mISDN/2-2, app-blacklist-check|s|1) in new stack
-- Executing LookupBlacklist(mISDN/2-2, ) in new stack
-- Executing GotoIf(mISDN/2-2, 0?blacklisted) in new
Hi Farook and the list,
You have may be forgotten to input that in the misdn.conf file :
nationalprefix=0
internationalprefix=00
dialplan=0
localdialplan=0
cpndialplan=0
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la
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