On 02:39, Fri 20 Oct 06, Tzafrir Cohen wrote:
On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote:
On 23:04, Thu 19 Oct 06, Vidar wrote:
Bristuff has been updated;
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
Thanks for the information.
It's a
I don't believe there is any quick simple way of doing this.
You would need to add the column in the DB and modify cdr.c.
I'm sure someone out there has a step by step doc on how to do this. You may try the #asterisk channel on irc.
bp
On 10/19/06, unplug [EMAIL PROTECTED] wrote:
Thanks!!Just
Doh! Turns out it won't be November. It will be a bit later. Sorry.
On Thursday 19 October 2006 21:35, Mike Diehl wrote:
On Thursday 19 October 2006 14:10, Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear
On 2006-10-19 20:30:03 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:
Hello, well, I need to configure two asterisk box like SIP trunks to se
nd sip
calls from one asterisk to the other and visceversa. So How I setup con
fi g
files to
Same similar problems here, with qozap and bristuff 0.3.0:
with physical media connected, I get a layer 1 down message
that keeps rolling up EVEN DURING AN INCOMING CALL on that BRI span,
and prevents asterisk from placing outbound calls...
until I restart asterisk (luckily, no kernel crashes so
On Thu, Oct 19, 2006 at 01:42:01PM +0100, Steve Davies wrote:
Hi,
For a significant time now (since about 0.2.0-rc8n) the qozap driver
has become very verbose if an ISDN line is not connected... I get the
messages below every couple of seconds in the asterisk logs.
Have you tried version =
Hi, I have just installed asterisk under Fedora5, the program runs fine except the color is not correct, it's very dark, difficult tosee. And if I select a part that part's color will be fine at least clear for read purpose. All I can do now is disabling color by changing the TERM environment
Hi all
I have downloaded 1.2.13
installing on my FC5
when iam making, iam getting the following error
could some one suggest me the what is the problem
make[1]: Entering directory `/root/vici/asterisk-1.2.13/apps'gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations
On Fri, Oct 20, 2006 at 03:34:32PM +0800, zuo bf wrote:
Hi,
I have just installed asterisk under Fedora5, the program runs fine
except the color is not correct, it's very dark, difficult to
see. And if I select a part that part's color will be fine at least clear
for read purpose. All I
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.
Cory,
OpenWRT -running on Linksys WRT-
Hi Doug,
I do not use extensions.conf so I cannot show anything but I can assure
that I do not set the callerid except for parameters inside zapata.conf:
usecallerid = yes
callerid = asreceived
Hope may help.
TIA
Giorgio Incantalupo
Doug Lytle wrote:
Giorgio Incantalupo wrote:
Hi,
I have
On 19 Oct 2006, at 21:10, Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users
embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of
the
email thread, if anyone is presently working with this scenario
please shoot
me an email.
I've
Hi Mitch,
I have same problemsometime I get that error in particular when I
modprobe module as root to fix asterisk wrong configurations but not
when rebooting. To be sure I chown /dev/zap inside my
/etc/init.d/asterisk launch script after modprobe-ing zaptel and wctdm.
Giorgio
Hi
Im very green to asterisk, and I have been asked if asterisk can be
used to do remote control, like opening gates etc, say when the user
dials a predefined number ...
And what hardware is required ...
Many Thanks
--
Gregory Machin
[EMAIL PROTECTED]
www.linuxpro.co.za
Dear folk,
My problem solved after two day research and try and error method ;). It was
related to rxgain of the board im using. I've set the rxgain to 12 and it
seems made some problem. As far as I got the spandsp is so sensitive about
noise on the line and because of that it couldn’t hand shake
Hi,
i have implemented Asterisk Realtime architecture
with Odbc and MySql DB. I have followed all the step of the documentation I
found on the Internet.
On the CLI, if I make "odbc show" I see that the DB
connection is "UP", but if I make "realtime load family column
value" both with
On 201006, 10:06, Giorgio Incantalupo wrote:
Hi Doug,
I do not use extensions.conf so I cannot show anything but I can assure
that I do not set the callerid except for parameters inside zapata.conf:
usecallerid = yes
callerid = asreceived
I guess the problem is at the telco's side, since
Oh this brings back memories.
- Original Message -
From: Andres [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, October 20, 2006 12:44 AM
Subject: Re: [asterisk-users] plainvoip - down ???
Joseph wrote:
Maurizio Pederneschi wrote:
Hi,
i have implemented Asterisk Realtime architecture with Odbc and MySql
DB. I have followed all the step of the documentation I found on the
Internet.
On the CLI, if I make odbc show I see that the DB connection is
UP, but if I make realtime load family
These are my conf file:
res_odbc.conf
;;; odbc setup file
; ENV is a global set of environmental variables that will get set.
; Note that all environmental variables can be seen by all connections,
; so you can't have different values for different connections.
[ENV]
INFORMIXSERVER =
I'm having the same issue overhere with a nearly identical config:
res_odbc.conf:
[mysql2]
enabled = yes
dsn = MySQL-asterisk
username = asterisk
password = asterisk
pre-connect = yes
extconfig.conf
[settings]
sipusers = odbc,MySQL-asterisk,sip_buddies
sippeers = odbc,MySQL-asterisk,sip_buddies
Hello,
I want to use Astribank from Xorcom,
has anybody some experience or references with it?
Sincerely,
Stepan
--
tel./fax: +420 552 305 306
email: [EMAIL PROTECTED]
www: http://www.ha-vel.cz
___
--Bandwidth and Colocation provided by
Hi
Im very green to asterisk, and I have been asked if asterisk can be
used to do remote control, like opening gates etc, say when the user
dials a predefined number ...
And what hardware is required ...
Many Thanks
Hi, yes it is possible using AGI scripts !
Greg
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi Jarkko,
I had the same problem..It worked with an old version of misdn-install
(taken from beronet site) but not with actual mqueue-misdn-install. I
tried to put it in every misdn.conf section I have without success. The
updated
Hi,I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement.
My friend dials 1-210-1234345, i want the asterisk to get
Can anyone point me in the direction of a good status
viewer for agents. I have looked at the voip-info wiki and saw some good
commercial ones. I just need opinions on any products. I am currently using FOP.
I am looking for login/out, ringing, hangup and the like. I do strictly monitor
agent
HiIm very green to asterisk, and I have been asked if asterisk can be
used to do remote control, like opening gates etc, say when the userdials a predefined number ...And what hardware is required ...We use it in a variety of situations in an industrial setting. We use it for some control, status
Hi all
after patching to my asterisk
when iam try to make, iam getting the following error
GNU_SOURCE -O6 -march=k8 -fomit-frame-pointer -fPIC -c -o app_voicemail.o app_voicemail.capp_voicemail.c: In function âsendmailâ:app_voicemail.c:1796: error: âVM_CONVERTMP3â undeclared (first use in this
Just adapt the Dial line you use like this
Dial(SIP/22${EXTEN})
On 10/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to
add 2 characters in front of the dialled number always when it send the call
to my asterisk.
- Original Message -
From: J. Oquendo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 19, 2006 3:22 PM
Subject: Re: [asterisk-users] plainvoip - down ???
Joseph wrote:
Is plainvoip down?
I've
M. Shokuie Nia wrote:
Dear folk,
My problem solved after two day research and try and error method ;). It was
related to rxgain of the board im using. I've set the rxgain to 12 and it
seems made some problem. As far as I got the spandsp is so sensitive about
noise on the line and because of
If you just want to control a couple of digital points this hardware
may be overkill, but it is cool stuff.
For smaller implementations you can just use the outbound control lines
(DTR RTS) on an RS232C port. That can give you control of two on/off
devices.
They only sink about 20ma so
Thank you for sharing this information .. Many Thanks , have a grate day :-)
On 10/20/06, marvin horst [EMAIL PROTECTED] wrote:
Hi
Im very green to asterisk, and I have been asked if asterisk can be
used to do remote control, like opening gates etc, say when the user
dials a predefined
Hi Tomislav,
may sound stoopid but have you checked if
/usr/lib/asterisk/modules/chan_misdn.so is present?
Giorgio Incantalupo
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi Jarkko,
I had the same problem..It worked with an old version of
Yeah, try the Flash Operator Panel. You can view/download it at:
http://www.asternic.org/
Good luck!
Jordan Novak wrote:
Can anyone point me in the direction of a good status viewer for
agents. I have looked at the voip-info wiki and saw some good
commercial ones. I just need opinions on
Thanks Tijl,That was a nice one but i would like to have my PAP2 programmed with this dialplan.This way i can program it for others PAP2's too. I want to have the PAP2 dialplan help not the asterisk dialplan. My PAP2 should send the 55-1-210-1234345 this way rather than 1-210-1234345.
ThanksOn
On 10/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Oct 19, 2006 at 01:42:01PM +0100, Steve Davies wrote:
Hi,
For a significant time now (since about 0.2.0-rc8n) the qozap driver
has become very verbose if an ISDN line is not connected... I get the
messages below every couple of
Wow, this is a completely neutral and very valuable review. Thanks a lot
Zoa.
I´m an * newbie; my little box will only needs 20 extensions to give
termination to remote users and I´m about to buy a PRI interface; I
decide to get Sangoma hardware.. a lot of people recommended it to me.
In
Friday, October 20, 2006, 2:09:56 PM, [EMAIL PROTECTED] wrote:
My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345.
Can someone help me to add this dialplan.
you must add :55 before every rule, where you want to add 55.
eg. this rule matches your example 1-210-1234345
On Fri, Oct 20, 2006 at 03:09:56PM +0300, [EMAIL PROTECTED] wrote:
I have a Linksys PAP2-NA connectd to my asterisk. I would like the
device to add 2 characters in front of the dialled number always when
it send the call to my asterisk. I dont know how to do that. I will
summarise
http://www.netphonedirectory.com/pap2_dialplan.htm
might help
remember google?
Bails
[EMAIL PROTECTED] wrote:
Thanks Tijl,
That was a nice one but i would like to have my PAP2 programmed with
this dialplan.
This way i can program it for others PAP2's too. I want to have the PAP2
On Thu, Oct 19, 2006 at 11:08:09PM -0500, Mitch Miller wrote:
* is having permission problems accessing /dev/zap/channel. When I
look, these devices (everything in /dev/zap) shows root.root for uid and
gid. If I start Asterisk from the command line, it runs fine (running
as Root). When I
Is anyone on this list familiar with Dallas? Anyone want to recommend something to
do on the Saturday/Sunday?
Never been to Dallas
so Im hoping for a restaurant recommendation for Saturday night.(somewhere
a little more up market would be good) also any sights that have to be visited
Hi,Thanks once again,Let me put is clear. I'm using TRIXBOX which many out here feel - its for kids- but i do like it an use it for sometime.As all i do have extensions and trunks configured on it. I want one of my extensions to use a particular outbound route only. I have rightly done all the
Title: Re: [asterisk-users] Astricon - post show Saturday?
I really recommend Pappas Brothers Steakhouse. My wife and I went there for our first anniversary, and it was really nice. Beyond that, check out the west end, its pretty nice place to be for the night scene.
On 10/20/06 8:45 AM, Dean
Hi list,
I have a client with a strange requirement: putting a noise gate on the
Asterisk channel. For those who are not familiar with them, noise gates
are used in musical instruments to avoid entering low-level noise into the
amp system. What they basically do is, they measure the volume of the
I've been using the Digium TE110P for more than a year now without any
hitches in a mission critical environment. It's in a Dell Poweredge
750.
I had a problem compiling libpri at first and the digium help desk was
more than helpful. I also had some echo issues, but those have mostly
been
The correct way to perform a factory reset on the Polycom phone is documented in one of our knowledge-base Articles, number: KB-001032http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-13.htmYou want to do a "factory format" to COMPLETELY erase everything on the phone.After
On 10/20/06, Steve Underwood [EMAIL PROTECTED] wrote:
M. Shokuie Nia wrote:
Dear folk,
My problem solved after two day research and try and error method ;). It was
related to rxgain of the board im using. I've set the rxgain to 12 and it
seems made some problem. As far as I got the spandsp is
Now I'm ready to begin playing with dial plans and am having a difficult
time getting started.
You may want to read the book :
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
That should help you
___
--Bandwidth and Colocation provided
On Fri, Oct 20, 2006 at 04:46:52PM +0300, [EMAIL PROTECTED] wrote:
As all i do have extensions and trunks configured on it. I want one of my
extensions to use a particular outbound route only. I have rightly done all
the setup in the extensions_custom.conf. Hence i do want the 55 to be
On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote:
Wow, this is a completely neutral and very valuable review. Thanks a lot
Zoa.
I´m an * newbie; my little box will only needs 20 extensions to give
termination to remote users and I´m about to buy a PRI interface; I
decide to get Sangoma
Tim Panton wrote:
On 19 Oct 2006, at 21:10, Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario
please shoot
me
Administrator TOOTAI wrote:
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please
shoot
me an email.
Cory,
Hi Jesse,
Do you know if latest bootrom
(3.2.2) and firmware (2.0.1) loads up onto Polycom IP500s?
Or
are they only for the later models?
Do
you know if you can still use TFTP for these software updates?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Dear all,
I've configured Asterisk Voicemail, but after some tests I realised that
when some call is sent to the voicemail of someone which username begins
with j letter, Asterisk gives me the error:
WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in
voicemail config file
Has anyone used the TrixBox/AAH builtin facility xPL for
facility (including home/office/industrial) automation?
On Fri, 2006-10-20 at 05:17 -0700,
[EMAIL PROTECTED] wrote:
Date: Fri, 20 Oct 2006 11:28:51 +0200
From: Gregory Machin [EMAIL PROTECTED]
Subject: [asterisk-users] using
Exactly what I was looking for. Thanks for the info. Going to go study
now ...
-- Mitch
Tzafrir Cohen wrote:
On Thu, Oct 19, 2006 at 11:08:09PM -0500, Mitch Miller wrote:
* is having permission problems accessing /dev/zap/channel. When I
look, these devices (everything in /dev/zap)
Hi guys,
This should be has an easy answer for you, my users are complaining
that when they press # and then ear gorgeous Allison Transfer the
timeout is very small, they must enter immediatly the extension to
transfer the call.
Is it possible to change this?
;transferdigittimeout = 3 ;
Hi Kristian,
http://www.voip-info.org/wiki/view/Mexuar
:)
see you at the show
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
Tim,
How do you use a web softphone
Thanks Steve, it was helpful to read your post. Neither Digium or
Sangoma single span cards have built in E/C, I´m wrong? To get one with
this feature enable is completely out of my budget; since it is no for
commercial use. I hope I wont have echo problems or they can be solve by
means of
Setup:
Asterisk server in NY.
Cisco 7960 IP Phones in NY and London.
Dedicated T1 from NY to Ldn.
T1:
Latency - 100ms
Qos applied
No errors
Default codec on Ldn IP Phones = g711alaw
Default codec on NY IP Phones = g711ulaw
Both codecs allowed on each phone.
Issue:
Calls on IP Phones from NY to
On Fri, Oct 20, 2006 at 04:46:52PM +0300, [EMAIL PROTECTED] wrote:
Let me put is clear. I'm using TRIXBOX which many out here feel - its
for kids- but i do like it an use it for sometime.
As all i do have extensions and trunks configured on it. I want one of
my extensions to use a
What version of * are you running? I have several j usernames in voicemail.conf under SVN-branch-1.2-r37458M. On 10/20/06, Ricardo Carvalho
[EMAIL PROTECTED] wrote:Dear all,
I've configured Asterisk Voicemail, but after some tests I realised thatwhen some call is sent to the voicemail of someone
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote:
Hi Jesse,
Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up onto
Polycom IP500's?
Or are they only for the later models?
Do you know if you can still use TFTP for these software updates?
They are compatible with
Dean Collins wrote:
Hi Kristian,
http://www.voip-info.org/wiki/view/Mexuar
:)
see you at the show
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
Arg!
I understand what a soft phone is. I
This might be a newbie question... I'm using a SIP trunk and trying
to get DID line information on an incoming call. All I hear is a
nice lady saying 'Zero' - then the call continues... Any suggestions?
thanks
Todd
exten = s,n,Set(DIDID=(${FROM_DID}))
exten = s,n,SayNumber(DIDID)
Kristian Kielhofner wrote:
Administrator TOOTAI wrote:
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario
please
Any news on this thread? I also need to know the way to get the R-URI
from sip INVITE messages received by Asterisk, through ${SIP_HEADER()}.
Thanks in advance,
Ricardo.
kjcsb wrote:
I have read the wiki about the SIP_HEADER function (http://www.voip-
Apparently it is down for GOOD :-/ (RIP)
http://www.voip-info.org/wiki/view/RIP+VOIP
--
#Joseph
On Fri, 2006-10-20 at 08:20 -0400, Mailing List wrote:
- Original Message -
From: J. Oquendo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'm currently running 1.4b3 with a Digium card and 23 g.729
licenses. Is there a way I can get the g.729 codec work off the CPU
and onto a DSP? Any T1/PRI cards with onboard codec DSPs?
-Matt
--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX
Just a wild idea:Store the filename in a variable before the call enters the queue - say RECFILENAME - and then once you know which agent has taken the call, execute an mv operation (using the system command) something like
system(mv ${RECFILENAME} ${RECFILENAME}-${AGENTNAME})i don't remember the
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
Through my testing I found
Subject: Re: [asterisk-users] Polycom boot error
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote:
Hi Jesse,
Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up
onto
Polycom IP500's?
Or are they only for the later models?
Do you know if you can
On Fri, Oct 20, 2006 at 11:54:45AM -0400, Kristian Kielhofner wrote:
I understand what a soft phone is. I know what java is. I also
know that neither have anything to do with the slug.
They do if Asterisk is runnin on the slug.
What he meant was perfectly clear to *me*,
Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote:
Sweet thanks for that, is there any reason not to go to version 2.0.1
now?
I know people were concerned initially because you cant go back but is
there a reason to go back if I have a few Polycom IP 500's?
We've got clients running 501's on
On Friday 20 October 2006 13:01, Matthew Crocker wrote:
I'm currently running 1.4b3 with a Digium card and 23 g.729
licenses. Is there a way I can get the g.729 codec work off the CPU
and onto a DSP? Any T1/PRI cards with onboard codec DSPs?
Digium's got their transcoder card. Are you
Dear Marco,
Take a look at featuredigittimeout, that might help :)
Regards.
---
M. Shokuie Nia
From: Marco Mouta [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
I playing a bit with this, it seems that if you use the new syntax it
works:
exten = _[a-z].,3,VoiceMail(${EXTEN}|u)
You can, of course, also use the b, j, s, and g flags.
Even using the VoiceMail(u${EXTEN}) still elides the 'j'.
Regards,
- Brad
-Original Message-
From: [EMAIL
On Fri, 2006-10-20 at 18:08 +0100, Ricardo Carvalho wrote:
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten =
I used to have fonality and I could press * when I got to someones voice
mail to go back to the menu. I assume I add that to the dialplan but
how? Thanks
BTW I went back to 1.2.12 and transfer works and DTMF works and it seems
to be much better for now.
Thanks for you help
Jason
Brian Capouch wrote:
Kristian Kielhofner wrote:
Administrator TOOTAI wrote:
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with
Jay R. Ashworth wrote:
On Fri, Oct 20, 2006 at 11:54:45AM -0400, Kristian Kielhofner wrote:
I understand what a soft phone is. I know what java is. I also
know that neither have anything to do with the slug.
They do if Asterisk is runnin on the slug.
What he meant was perfectly clear
We are having an issue with transferred calls being dropped.
Looking at the asterisk 1.2.10 logs, it appears that when it is dropped,
the SIP unit send a CANCEL message to the server.
On successful transfers this is not seen.
The errors logged in the SIP Unit error log, I believe are from
Hi Steve,
As a matter of fact, you've done a greate job in writting this library, no
doubts. I really dont know rxgain = 12 makes that much distortion but I'm
curios to know if I pass through the incoming fax to an analog fax machine
on another fxs line, the machine wouldn't receive the fax
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers
control to the context defined in the IAX.conf -where I have the
trunk line defined...
On 20 Oct 2006, at 19:05, Kristian Kielhofner wrote:
It actually had *nothing* to do with Asterisk running on the slug,
so it seems that you might be even more confused than I am :). He
confirmed off-list that the scenario he described did not involve
running Asterisk on the slug. It
I am attempting to use txfax on centos 4.4
the libtiff is:
libtiff-devel-3.6.1-12
libtiff-3.6.1-12
Is this OK or do I have to download the libtiff stuff and install it also.
I am not having much luck faxing yet. I receive 1/3 pages or 2/3 pages
etc...
I have yet to receive 3/3 pages.
Tim Panton wrote:
On 20 Oct 2006, at 19:05, Kristian Kielhofner wrote:
It actually had *nothing* to do with Asterisk running on the
slug, so it seems that you might be even more confused than I am :).
He confirmed off-list that the scenario he described did not involve
running
On Oct 19, 2006, at 3:00 PM, [EMAIL PROTECTED] wrote:Date: Thu, 19 Oct 2006 09:30:38 -0500 From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello all!
I have a few problems with Snom 320 phones:
Problem A - Transfer out of Queues:
We have a call center with some Snoms. We are using Queue and
AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer
a call out of the queue using the hold and transfer buttons
on the Snom. This
Hi,
I've install zaptel and I don't have a Digium card installed in the machine.
So I want to install ztdummy to have Music On Hold working. I've follow
these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and
at the point modprobe ztdummy it's failing. I'm getting these
This might be a newbie question...
You're right, part ofit is. I don't mean to sound rude, but you really need to go do some research first to get the basics down. First place is to read the book, Asterisk: The Future of Telephony (available for free, there's this site called
google.com that
HK == Håkan Källberg [EMAIL PROTECTED] writes:
HK Problem B - Quick Dial Buttons:
HK I have used the programmable function keys together with the hint
HK system in * to monitor local lines. It works very well,
HK impressive! But people like to use these buttons as quick dial
HK buttons for
Here you go, from the voip-info.org wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail
Also. during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail
context. This needs an example
'#' - the greeting and/or instructions are
Top posting since this is simple. Your kernel does not have
a RTC compiled in or as a module...
Do you build your own kernels? If so add RTC as a builtin or
a module. If you use a distro kernel, you might be able to
modprobe the RTC module.
Dan
-Original Message-
From: [EMAIL
Has anybody out there, on non-FPU embedded platorms, made any good use
of things like ilbc and Speex?
The exisiting implementations of both run very poorly on a non-fpu cpu's,
especialy if clock speed 400 Mhz
I have run asterisk (and still do) on mips,ixp and powerpc (all without
fpu's) and i
Todd- Asterisk wrote:
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers control
to the context defined in the IAX.conf -where I have the trunk
On Fri, Oct 20, 2006 at 04:51:47PM -0400, Jean-Etienne Kelly wrote:
Hi,
I've install zaptel and I don't have a Digium card installed in the machine.
So I want to install ztdummy to have Music On Hold working. I've follow
these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
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