hi all,
i m using debian to run my asterisk
gateway.I want to make some customization in voicemail
application.For that i need to modify voicmail's
.C(source file) file. can any body tell me where
exactly all .C files resides in the system..
thanks
tejas
I have used both Telular analog units and Voiceblue SIP units in Australia.
PaulH
- Original Message -
From: Adrian Carter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, January 07, 2006 1:40 AM
Subject: Re:
On Sat, 2006-01-07 at 00:26 -0800, Tejas Shah wrote:
hi all,
i m using debian to run my asterisk
gateway.I want to make some customization in voicemail
application.For that i need to modify voicmail's
.C(source file) file. can any body tell me where
exactly all .C files resides
There is a sample php script in the contribs folder
that shows who is logged in - one of my clients uses it.
PaulH
- Original Message -
From:
Dov Bigio
To: asterisk-users@lists.digium.com
Sent: Saturday, January 07, 2006 8:24
AM
Subject: [Asterisk-Users]
On 6 Jan 2006, at 16:28, Joan Bautista wrote:Hi, I haven't found anything about the message below on the mailing list, Does anyones knows why this notice is being appearing? -- Executing Dial("Local/[EMAIL PROTECTED],2", "IAX2/CallOut/12365533643|30|otT") in new stack -- Called
I still having problem with remote SIP client,
trying to use IAX client instead but i've to
specify TCP port 8080 (because of firewall).
I did this on server in bindport=8080 in iax.conf
but i cannot find a soft client that allow to set wich
server port to use.
Any idea?
Thanks, Antonio
I am having the same problem with a male voice at the other end.
It is making the spa3k problem for me.
Has this been reported to SIPURA ?
Is this a common problem ?
has anyone done been able to make this happen less often ?
I would hope perhaps there's some kind of setting
Hello,
I'm traying to link 2 * servers using SIP and the
following errors was show:
"SIP/AsteriskA:[EMAIL PROTECTED]/100")
in new stackDec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No
such host: 10.0.0.121/100Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759
dial_exec: Unable to
I'm in conversation with Draytek's pre-sales dept..
Here's the most recent reply:
Hello,
We really don't know of anyone who has run an Asterisk server on
a Vigor2900. There are doubtless people around, but it's relatively
rare. Most people don't run SIP servers.
Regards,
All I
Jonathan Attwood wrote:
I'm in conversation with Draytek's pre-sales dept..
I bought a 2600 2 years ago and I had alot of NAT problem, because the
SPI was changing the externhost (sip.conf) ip address with the local
private address forwarding the packets, so the audio stream
Hi,
Draytek 2900 is a great router. Easy to setup stable. I want known more
detail of your network configuration. I can setup it and make some test.
Regards,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Attwood
Sent: Saturday, January
Now draytek have some SIP embeded router (e.g., 2100VG, 2900VG...). Maybe
you can try these new router.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergio
Chersovani
Sent: Saturday, January 07, 2006 9:16 PM
To: Asterisk Users Mailing List -
Try this.
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio Gallo
Sent: Saturday, January 07, 2006 8:20 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] wich IAX soft
No Bluetooth in the Samsung T309. I couldn't think of why I'd want
BT... then of course I started looking at cell sockets, etc. after I
got it and found several do not have a cable for the T309 yet. In
hindsight, bluetooth would have made this easier. Live and learn!
On 1/6/06, Jonathan Attwood
Yes, I would be very interested in this as well.
Thanks,
Steve
-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, January 06, 2006 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialer
Very cool! Is
Darren,
I am interested in your project. Let me know if I can help you test.
Thanks,
Steve
-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, January 06, 2006 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
On Sat, Jan 07, 2006 at 01:19:34PM +0100, Antonio Gallo wrote:
I still having problem with remote SIP client,
trying to use IAX client instead but i've to
specify TCP port 8080 (because of firewall).
The IAX protocol is based on UDP, not TCP.
I did this on server in bindport=8080 in
-- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack
I think the problem here is that you have the timeout set to one second
and I am not sure what the * is before 101.
My interpretation is ext-local specifies local context. *101 means
dial extension 101 but I am unsure of what the * is
Hi,
I having similar problem. Unfortunately each thread is archive leads to
nowhere. I read a post in which similar problem was solved by changing rxgain
and txgain to 15. Maybe this would help.
Does anyone have common problems?
I was wondering why asterisk - great telecommunication program -
Hi,
I have a problem with pattern matching N what should digit 2 to 9
in Asterisk 1.2.1.
If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the
context dialout and find there an matching _2. and is using this.
If I change _NNN to _XXX everything works fine. If I dial 220
I added:
mailcmd=/usr/bin/sendmail -f hostname -t
to the voicemail.conf file under [general]
- James
On Jan 6, 2006, at 10:31 PM, Pisac wrote:
Yes, I found that this is problem with my server. Second server is
connected through second provider, and first server and my domain is
hosted at
network problems. Asterisk wan unable to connect or bind to 10.0.0.121/100
Regards
On 1/7/06, Cleyverson P. Costa [EMAIL PROTECTED] wrote:
Hello,
I'm traying to link 2 * servers using SIP and the following errors was show:
SIP/AsteriskA:[EMAIL PROTECTED]/100) in new stack
Dec 13 22:46:57
andrutto wrote:
I was wondering why asterisk - great telecommunication program - has such a
weak fax support.
Because it's a PBX and not a fax server.
Use IAXmodem and HylaFAX, and then you have a fax server.
http://sourceforge.net/projects/iaxmodem
http://hylafax.sourceforge.net/
Lee.
On 07/01/06, Thomas [EMAIL PROTECTED] wrote:
Hi,
I have a problem with pattern matching N what should digit 2 to 9
in Asterisk 1.2.1.
If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the
context dialout and find there an matching _2. and is using this.
If I change
andrutto wrote:
Hi,
I having similar problem. Unfortunately each thread is archive leads to
nowhere. I read a post in which similar problem was solved by changing rxgain
and txgain to 15. Maybe this would help.
Does anyone have common problems?
I was wondering why asterisk - great
Jonathan Attwood wrote:
I'm in conversation with Draytek's pre-sales dept..
Here's the most recent reply:
Hello,
We really don't know of anyone who has run an Asterisk server on
a Vigor2900. There are doubtless people around, but it's relatively
rare. Most people don't run SIP
thanks...
_NXX works for me
best regards
Thomas
On Saturday 07 January 2006 16:37, Peter Bowyer wrote:
On 07/01/06, Thomas [EMAIL PROTECTED] wrote:
Hi,
I have a problem with pattern matching N what should digit 2 to 9
in Asterisk 1.2.1.
If I dial 220 I did not get an PlayBack
Lee Howard [EMAIL PROTECTED] wrote:
Use IAXmodem and HylaFAX, and then you have a fax server.
http://sourceforge.net/projects/iaxmodem
http://hylafax.sourceforge.net/
If you're looking for more general information on HylaFAX, see
www.hylafax.org.
-Darren
It certainly does.
How many rules can you create in the port forwarding section of the V2900?
I was told that the V2900 has SIP_ALG. Is this something you've activated?
On 1/7/06, Faris Raouf [EMAIL PROTECTED] wrote:
Jonathan Attwood wrote:
I'm in conversation with Draytek's pre-sales
I am working on a project to unite several local school districts. We will
have 14 different districts, every district would have their own asterisk box
on location. We all have fiber lines running to a central location at our
isd. This provides connectivity to all the districts.
1.
I having similar problem. Unfortunately each thread is archive leads to
nowhere. I
read a post in which similar problem was solved by changing rxgain and txgain
to 15.
Maybe this would help.
Does anyone have common problems?
I was wondering why asterisk - great telecommunication program
Bill Michaelson wrote:
I am having trouble understanding how to use this. I want to transfer
certain incoming calls from an IAX ITSP based on caller ID. From what I
can make of the docs, I thought I need to do something like this...
exten = _NXXNXX,n(nocid),transfer(1000)
exten =
Hi,I certainly don't want to integrate fax-e-mail support into spandsp.I think our problem is not connected with spandsp and fax - email integration. All the applications I mean spandsp txfax and rxfax are enough to have emial - fax functionality in Asterisk. I wrote a program which allows me to
Any type of circuit available as an analog line is also available
over a T1. It just minimizes the amount of copper required to deliver
service.
You must look at you original order from your telephone company to
determine the type of circuits they are delivering. They may be POTS
1FB in
Do not know what version you are
running,
But there are a few ways to do
this.
There is a persistant setting:
from agents.conf
;; Define whether callbacklogins should be stored in
astdb for; persistence. Persistent logins will be reloaded after;
Asterisk restarts.;persistentagents=yes
If
I would agree with all but a few issue:
I would incoparate dundi, After using it I have fallen in love with it
for distributed applications such as this. It makes configuration at
first a bit steeper but it picks up monentum as your deploy. With Dundi
you basicaly broadcast what extensions or
That would depend heavaly on your netowrk. Would your Swtiches (not
routers as TMDoE is layer 2) I pulled up an old posting from Mark on
TDMoE.
http://www.marko.net/asterisk/archives/0301/0566.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
A (too) simple sollution to your problem is to take the analog audio from
your IP phone using a module atached between the curly handset cord and
the base unit of the IP phone - like
http://www.quasarelectronics.com/tre156.htm
So, basically you need to change the old RJ11 - 1/8 inch recording -
http://www.astpp.org/index.php?n=Misc.AutoDialOut
I put together what I have on that site.
Darren wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
Darren,
I am interested in your project. Let me know if I can help you test.
Thanks,
Steve
-Original Message-
From: Wiley Siler
Rich Adamson wrote:
Its a fairly common problem with the spa3k, but is somewhat dependent on
how it is configured and the distance to your CO.
Just to add a couple of data points:
I don't know why, but for me the problem has been worse lately than it
had been during the early time I spent
Of course that's not a problem to
use hylafax, but I just want to have it on one machine (I'm afraid that Asterisk
and hylafax won't run on the same
machine :( )
[Colin
Anderson]I am experimenting with IAXmodem to Hylafax running on an
Asterisk server. It works. Last Thur, I had 98 virtual
Hello All,
Dunno what happen but Asterisk is refusing to start... Went over the
log and found out that app_rxfax.so is failing to load.
Jan 7 11:57:28 VERBOSE[4320] logger.c: [app_rxfax.so]Jan 7
11:57:28 WARNING[4320] loader.c: /usr/lib/asterisk/modules/
app_rxfax.so: undefined symbol:
Yes, you could do that making some changes on modules.conf
noload = app_rxfax.so
Regards
Alberto
Nitesh Divecha wrote:
Hello All,
Dunno what happen but Asterisk is refusing to start... Went over the
log and found out that app_rxfax.so is failing to load.
Jan 7 11:57:28 VERBOSE[4320]
From this server can you ping 10.0.0.121?
What is your network mask?
10.0.0.121/100 is not a valid address (mask are in the range of /0 to /32)
This is where you should start.
What is your network definition?
Tudo bem?
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
Hello to all
I do not know what is causing choppy music on hold
when call comes in through E1 card (PRI).. but this channel info is somehow
strange.. We use Alaw over PRI (and I think its format number 8),
But why is WriteFormat at 2 ?
Thanks!
show channel Zap/1-1
--
I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the
range of 0600 - 1699. However, the second that the 0 is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call. My 16xx numbers wait for all four
digits before trying
Ken D'Ambrosio wrote:
I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the
range of 0600 - 1699. However, the second that the 0 is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call. My 16xx numbers wait for all
Running a fairly recent subversion release of Asterisk, I'm running into
a problem using labels (as opposed to priorities) with this application.
Here is the dialplan segment:
; isolate gotoiftime bug with labels
;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4)
exten =
I'm curious why the number of jobs out there requiring Asterisk seems to be
pretty low. After looking around dice, monster, careerbuilder etc, I was
surprised to find no more than 3-4 employment opportunities with Asterisk
throughout the US.
Is it really that low? There seems to be a job of
Hiya,
I've got a 2900g series, and it works fine (I have the 2200we before I
upgraded and that was ok too!). I have used its built in wifi to go to
an ipaq iax extension, and also have asterisk doing sip and iax through
to fwd and sipgate. There's some port forwarding rules to get the
Bill Michaelson wrote:
Running a fairly recent subversion release of Asterisk, I'm running into
a problem using labels (as opposed to priorities) with this application.
Here is the dialplan segment:
; isolate gotoiftime bug with labels
;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4)
Hi,
I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7
running on the same box.
I wonder how to setup the iax account correctly so that I may
initiate outbound calls via iaxmodem?
registration upon iaxmodem startup is okay and I can direct calls to it.
-- Registered IAX2
[user[:secret[EMAIL PROTECTED]peer[:portno][/exten[@context]]
Well but i don't need to dial out, i need to register to asterisk
using IAX and 8080 port and all the client i've tested will not
allow that into their account config section: they just have the server
name/ip not the port.
Bruno Voigt wrote:
Hi,
I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7
running on the same box.
I wonder how to setup the iax account correctly so that I may
initiate outbound calls via iaxmodem?
registration upon iaxmodem startup is okay and I can direct calls to it.
Most of the Asterisk work I have found out and about is either done by
internal staff or by companies wanting work done by external contractors.
Like you, I have found very little in the way of full time jobs for
'asterisk people'
PaulH
- Original Message -
From: Douglas Garstang
You could probably pay $15-20 for a paul budde report with relatively
accurate figures. www.budde.com.au (even if he does believe asterisk is
a passing fad - hi Paul :) he's still one of the best telco resources in
Australia.
Telsyste might be another option.
Cheers,
Dean
-Original
Redhat has a 'Hardware Discovery Utility' called
Kudzu.
When I change cards, kudzu pops up and ask to
remove/config the card.
Most of the time kudzu has trouble recognizing the
Digium Zaptel cards and calls them something wrong, like calling the TDM card a
network card.
I'm having a
Hi,
Some background... I have the following directories:
/var/lib/asterisk/sounds/custom/ - Here are french prompts
/var/lib/asterisk/sounds/custom/en - Here are the english prompts
If I do:
SetLanguage(en)
Playback(custom/mypromp)
The prompt file is played
Asterisk is still virtually unknown to endusers. The only reason why
it's even a blip on the radar of PBX manufacturers is because how
quickly the community is growing, and how feature rich the system is
already. The biggest threat is that it is free and not proprietary
which totally flies in the
Thanks
Sorry, I missed that local/8600 channel.
On 1/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote:
Hi!
Thanks for that post thats a good one
:-)
just one thing, what happens if the user doesn't want to connect to the
caller? does it get saved as VM? Looking thru the code I
Douglas Garstang wrote:
I'm curious why the number of jobs out there requiring Asterisk seems to be
pretty low. After looking around dice, monster, careerbuilder etc, I was
surprised to find no more than 3-4 employment opportunities with Asterisk
throughout the US.
Is it really that low?
Would like some advice on the best way to route DID's to remote
asterisk servers. Currently I have multiple DID's on my main Asterisk
server in a datacenter and have remote servers that connect via an IAX
trunk and when a call comes into my server I pass it to the iax peer.
Just wondering what
I have written an agi script that I use for that. Then I can just have
a list of dids and extensions in a db.
Tom Vile wrote:
Would like some advice on the best way to route DID's to remote
asterisk servers. Currently I have multiple DID's on my main Asterisk
server in a datacenter and
It's funny you mentioned that Darren, I was looking at your scripts
today. I will evaluate it some more.
On 1/7/06, Darren Wiebe [EMAIL PROTECTED] wrote:
I have written an agi script that I use for that. Then I can just have
a list of dids and extensions in a db.
Tom Vile wrote:
Would
Post your extensions.conf and what's on the CLI (asterisk -r)
As requested:
# cat /etc/asterisk/extensions.conf
[incoming]
exten = s,1,Answer()
exten = s,n,NoOp(CallerID is ${CALLERID})
exten = s,n,NoOp(DID is ${DNID})
exten = s,n,Background(enter-ext-of-person)
exten =
On 1/7/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Post your extensions.conf and what's on the CLI (asterisk -r)
As requested:
# cat /etc/asterisk/extensions.conf
[incoming]
exten = s,1,Answer()
exten = s,n,NoOp(CallerID is ${CALLERID})
exten = s,n,NoOp(DID is ${DNID})
exten =
Do I need to install the complete ASTPP package or just utilize your
AGI script with the context for AMP?
Thanks
On 1/7/06, Tom Vile [EMAIL PROTECTED] wrote:
It's funny you mentioned that Darren, I was looking at your scripts
today. I will evaluate it some more.
On 1/7/06, Darren Wiebe [EMAIL
I am stumped as well, you don't have any extension defined for either
0, _0X, or _0X.
So I got no clue why *you* are stumped, in fact 1625 is treated
special, because it got an extension.
Okay; thanks! I mis-understood the mechanism. I didn't think
extensions.conf actually came into play
Just grab the script. I can help you with it off the mailing list if
you like
Darren
Tom Vile wrote:
Do I need to install the complete ASTPP package or just utilize your
AGI script with the context for AMP?
Thanks
On 1/7/06, Tom Vile [EMAIL PROTECTED] wrote:
It's funny you mentioned
Hi all,
Can anyone tell me from where i can call my update query when the call
is pickuped using AGI Perl.
I writen the code to store active calls in MySQL DB. and display the
real time call counter. i mean to insert record when the call is
pickuped
--
Thank You,
Code Lover
Thanks all for the replies. I started working for a CLEC a few months ago and
we've chosen to implement Asterisk. I'm not sure if the fact that my boss is an
open source advocate is a good thing or not... ie yes it's great to work with
Asterisk and see all the features coming together
I have been trying to figure out for quite a while now how
to better setup asterisk in a small office environment.. For example,
small offices usually want to be able to have shared lines, so one can put a
line on hold and another person can pick that call up if its on
hold. The astra and
On 1/8/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm not sure if the fact that my boss is an open source advocate is a good thing or not... ie yes it's great to work with Asterisk and see all the features coming together (especially with Polycom phones). On the other hand I wonder how useful
I've been Googling around for some time now (a few
hours on dial-up). I find all kinds of questions similar to mine, but
either there is no answer or the answer has nothing to do with the
question. Hopefully this post isn't another one of those.
Does Asterisk favor FPU performance or clock
On 12/27/05 22:39 Kevin P. Fleming said the following:
Steve 4 wrote:
Field-upgradeable? Does that mean that I can do it myself? That would
be great since some systems are in production and sending the board to
Digium takes time.
The 2nd gen firmware has field-upgradeability. The 1st
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