In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> what about this?
> show app ringing?
>
> exten => _7XX,1,Ringing
> exten => _7XX,2,Goto(local,${EXTEN},1)
It looked promising so I tried it. Unfortunately it didn't help. Calling person
doesn't hear ringing. I don't know why this applic
Anyone can help me to solve the problem about playing the prompt? Is
it related to the package problem? Anyone can give me a clue to find
out the solution? Thx.
I have a simple dial plan to play a voice prompt as follow.
exten => ,1,Answer()
exten => ,2,Playback(you-have-reache
Hi users;
i am new in the mailing list and asterisk user . i
have to implement METHOD 3 of the link
(http://www.voip-info.org/wiki/view/Asterisk+at+large&view_comment_id=11963)
i have question that is:
Q:when lets i have getting a NOTIFY message and my
phone changes the tone to a MWI tone now if
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Hi all,
>
> the default message for email notification looks like:
>
> Is there something wrong with my config?
> thx in advance
This should work. Have you reloaded Asterisk?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000
the situation
Asterisk <-- SIP ---> SIPGW <--- SIP Phone
SIP Phone is trying to call asterisk dialplan:
exten => 0224577501,1,Answer()
exten => 0224577501,2,Playback(demo-instruct)
exten => 0224577501,3,Hangup()
however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a "488 Not
acceptable her
Brandon Galbraith wrote:
The problem is not present in AMD systems though, correct?
-brandon
stop top-posting ! :D
[snip]
those have intel processors inside, no ?
you may be a victim of the bus sharing crap that intel uses (that is,
processors are in parallel on the processor =
The problem is not present in AMD systems though, correct?-brandonOn 9/19/06, Raphael Jacquot <[EMAIL PROTECTED]
> wrote:George Pajari wrote:> Any thoughts on this one?>> IBM xSeries 330 processor running Debian
3.1 (2.6.8 kernel) with a> TE406P board.>> Working fine (more or less) connected to a
George Pajari wrote:
Any thoughts on this one?
IBM xSeries 330 processor running Debian 3.1 (2.6.8 kernel) with a
TE406P board.
Working fine (more or less) connected to a couple of PRIs.
Rebuild kernel with support for second CPU and inbound (PRI -> SIP)
audio is badly garbled. Outbound (As
Douglas Garstang wrote:
I wonder if the look and feel of this GUI will be completely configurable. If
it's not, then I really don't think that's very useful. Service providers
wouldn't be able to use it to let their customers manage their own settings,
and customers wouldn't want to use it if
I am thinking if re-invite will interfere accounting.
Please help me to figure it out:
Phone A is registered at asterisk and calls a gateway. If the gateway
allows re-invite than the rtp would go directly from phone A to the
gateway, while the sip messages are still going through Asterisk.
As
IAX has some pretty severe limitations when it comes to trunking calls between
Asterisk boxes. It can't pass variables for example, and any calls to SIP
phones at the far end will be treated as IAX calls, which is just nuts. This
means you lose a lot of SIP features, like transferring and forwar
I wonder if the look and feel of this GUI will be completely configurable. If
it's not, then I really don't think that's very useful. Service providers
wouldn't be able to use it to let their customers manage their own settings,
and customers wouldn't want to use it if it wasn't branded with the
Yes, it initiates a completely new call. No sending DTMF in the current call
I'm afraid.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Sun 9/17/2006 2:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
48 was the limit on the number of speed dial entries that you could have in the
directory. 7 was the old limit for the number of buddies you could watch. As
far as I know, in 2.0.1, the number of entries you can have in the speed dial
directory is 99, and the number of buddies that you can watch
Hil Folks,
I am trying to use latest spandsp(0.0.2) with asteirsk 1.2.9.1
and tiff library 3.7.1 through a SIP channel but the channel is freezing
after "answer". It was running ok using 1.0.10.
Any tipo to make it work well?
Thanks in advance,
Isamar
Versions 1.6.x supported NAT as well.
-Original Message-
From: Matt Florell [mailto:[EMAIL PROTECTED]
Sent: Mon 9/18/2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Polycom Soun
So the press announcement said that the new Digium GUI will be available in
v1.4 sometime in Oct. Is the GUI already there in Trunk or is there some
other branch of development that the general public cannot access?
Do you mean this?
http://svn.digium.com/view/asterisk/trunk/static-http/
On
And also it would be nice to have more detailed instructions on how to upgrade FOP.
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I do use FOP for myself, and can also do some work in HTML, but really don't have time to modify it for someone. Thats why I was thinking of HUD. Otherwise FOP has more features than HUD Lite.
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as
Actually when I checked it tonight again, it was back to normal. I really didn't understand how and why. But as long as it is working now, I'll not touch anything again. But I really wanted to upgrade to FOP 2.6, how can I do that?
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That's all well and good, but there are some phones out there that pack
samples into RTP payloads using the AAL2 direction. This causes interop
nightmares (i.e. your phones talk G.726-32, someone elses phones talk
G.726-32, but it sounds rubbish when you attempt a conversation). I
would guess that
I use two user's per host one for user and the other peer. Sort of
like attahed.
I also prefer IAX for communication between asterisk boxes. IAX use's
less bandwidth than SIP and it's trunks are alot smaller. If you look
at SIP traffic, 80% of it is headers. The headers look just like smtp
he
Thanks I'll give them a trial.
-- Original message -- From: "Insider KT" <[EMAIL PROTECTED]> > I've used this company now for over a year. > It is part of Ipcb.net, so you got live support 24 hours a day every day. > The quality is very good and the reliability is near pe
I've used this company now for over a year.
It is part of Ipcb.net, so you got live support 24 hours a day every day.
The quality is very good and the reliability is near perfect. You can have
1000 simultaneous calls.
On the down side - The Signup is not so easy. I had to fax 7 papers to
verif
The only load I have is,
load => chan_modem.soload => res_musiconhold.so
[global]chan_modem.so=yes
-- Original message -- From: "Justin Tunney" <[EMAIL PROTECTED]> > Check /etc/asterisk/modules.conf and see if there is a line trying to load it. > > On 9/18/06, [EMAIL PR
I have created a context in extensions.conf and when I dial, it is suppose to ask me to enter pin number but instead this the error I get.
Sep 18 18:11:54 WARNING[6514]: chan_sip.c:1968 create_addr: No such host: 4035Sep 18 18:11:54 NOTICE[6514]: app_dial.c:1011 dial_exec_full: Unable to create
Check /etc/asterisk/modules.conf and see if there is a line trying to load it.
On 9/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
[EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk start
Starting Asterisk PBX: FATAL: Module ixj not found.
__
When I started Asterisk I get this error but it is working fine and should I be concerned. Error below:
[EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk startStarting Asterisk PBX: FATAL: Module ixj not found.
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Just wondering about the Asterisk Appliance. I was waiting for hardware
like this but the details are still kind of sketchy about how it will be
sold. Will there be an option to buy it barebones without Asterisk Business
Edition?
Not even sure if it's feasable for a mere mortal such as mysel
Any thoughts on this one?
IBM xSeries 330 processor running Debian 3.1 (2.6.8 kernel) with a
TE406P board.
Working fine (more or less) connected to a couple of PRIs.
Rebuild kernel with support for second CPU and inbound (PRI -> SIP)
audio is badly garbled. Outbound (Asterisk -> PRI) is fine
You mean the menuselect ncurses screen? If yes, then yes... it's a gui. :)
-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED]
Sent: Monday, September 18, 2006 4:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Digium GUI?
So the press announcement said that
So the press announcement said that the new Digium GUI will be available in
v1.4 sometime in Oct. Is the GUI already there in Trunk or is there some
other branch of development that the general public cannot access?
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Hi,How would you best learn VoIP Quality of Experience ?Before diving into packet loss and jitter, I would like to know what a toll-quality call is, what a rated 3.5 MOS call is like.I'm wondering how I should proceed.
Shall I :- get pre-recorded sound files somewhere and simply stream them to a MO
Hi everybody,
I'm trying to use the periodic-annouce and periodic-announce-frequency
options.
I use Asterisk 1.2.9.1-BRIstuffed-0.3.0 with MySQL realtime
configuration. It seems that Asterisk Realtime queues doesn't support
these options.
When I try to add the 2 fields to the MySQL table, not
Mike,
If you could send the answer here it would be greate. I would like to add
this features to my Polycom phones too.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Anthony
Rodgers
Envoyé : 18 septembre 2006 19:09
À : Asterisk Users Mailing Lis
Are you having this problem with an analog line or PRI ?
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tobias Wolf
Envoyé : 18 septembre 2006 11:41
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Dial and Time
Hi Mike,
It's done using the digitmap feature of sip.cfg - email me offlist or
come on #asterisk and I can help you with the specifics.
CP
On 18-Sep-06, at 11:08 AM, Mike wrote:
I'm trying to make the Polycom 501 go off-hook (in speaker phone
mode) when any digits is dialed and the handse
Centos is a free version of RHEL.
Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Monday, September 18, 2006 12:40 PM
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll
work on my Asterisk box (outside of the FXO & FXS modules on the TDM card in
the Asterisk server, I only run SIP on the hardphones).
I don't know the phone's password (sound familiar?). - Have tried
everything, "cisco", "
bilal ghayyad wrote:
> Hi list;
>
> Does asterisk work with fedora because redhat
> enterprise is licensed and costly.
>
> Regards
> Bilal
>
> __
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yaho
Hi,
we're using Fedora Core 3-5 on all of our customer asterisk installations,
as well as on other projects like mythtv, mailservers and general servers
without any problems.
We like Fedora's bleeding edge state.
Guido
>
> Hi list;
>
> Does asterisk work with fedora because redhat
> enterpri
You'll need to change your XP box to the default router of the phone (or, just change your XP to a static on your local net, then add a static ip under advanced for the default route of the phone, that way you still have access to the internet from your XP box), and then add (under advanced) the IP
Ok. First question is how to make it say my number back.
Like if you call extension 1000 from extension 1001, I want
it to say “Number is 1,0,0,1” like an ANI number? Help.
Also I want to setup a meetme conference so that it asks “Enter
conference number” then execute meetme($entere
Hi,
You might need to recreate fop config files with amp tools. The script
that does that is named retrieve_op_conf_from_mysql.pl. You might also
have permissions problems on the directory or files inside them. They
must be owned by user asterisk if I recall correctly. Good luck,
On 9/18/06, Zee
Just curious how most of you are defining SIP peers in
sip.conf – for Asterisk boxes talking to each other. Are most of
you just making a type=friend connection and a single context or are you
separating them out to in/out definitions and contexts?
In other words
Where voicegw1 is the
Hi Rick,
Am Montag, 18. September 2006 21:30 schrieb Rick Smith:
> can't the agent just transfer the caller to another extension, whether that
> be another queue or a person ?
>
yes, that's the easy part. But my client wants the caller (!) to be able to
transfer himself into another context. The
Try to decrease the gain value on the Audiocodes.
That solved the echo issues we had.
On 9/18/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
I have installed some AudioCodes analog gateways in conjunction with * and am having an annoying problem on the mp-118-fxs side. call quality is decent
Cory
That sounds like "glare" and could be caused by the AudioCodes. Check
your amplification or line power as I am sure you are not powering 5
miles of copper.
On 9/18/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
I have installed some AudioCodes analog gateways in conjunction with * and
am
I have installed some
AudioCodes analog gateways in conjunction with * and am having an annoying
problem on the mp-118-fxs side. call quality is decent but on the mp118 you
hear yourself in your ear with around 1/10 to 1/4 of a second delay. this is
very annoying and makes it difficult to c
can't the agent just transfer the caller to another extension, whether that be
another queue or a person ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan-Michael.
Guenther (in-put GbR)
Sent: Monday, September 18, 2006 3:19 PM
To: asterisk-users@l
I am looking to setup asterisk on a DRBD distributed volume. The
overall goal is to setup a redundant server that will kick in when the
primary fails. So I am thinking of just setting up a directory like
/drbd (or /mirror) and install the entire asterisk application (logs,
var run, etc) all on /
Hi,
I would like to give a caller the chance to leave a queue after an agent has
already accepted the call.
The caller enters the queue by dialing 333:
[from-sip]
exten => 300,1,Answer()
exten => 300,2,Queue(q1|tT)
When the caller presses # and e.g. 1, asterisk is looking for this extension
i
- Original Message -
From: "Paul Hewlett" <[EMAIL PROTECTED]>
To:
Sent: Sunday, September 17, 2006 3:34 PM
Subject: Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 -
problemwith module versionmagic
On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote:
On Fri, Sep 1
I'm not anything to do with them, but sounds a nice design.
CSR have introduced a VoWiFi reference design that costs around $20.
The interesting thing is that it supports both SIP and IAX2.
Maybe Digium should make a WiFi handset ...
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)2
On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote:
> Hi list;
>
> Does asterisk work with fedora because redhat
> enterprise is licensed and costly.
>
check out Centos.
> Regards
> Bilal
>
> __
> Do You Yahoo!?
> Tired of spam? Yahoo
I have been using asterisk on FC4, FC5 and now FC6t3 with no irritating problem.
- Original Message -
From: "bilal ghayyad" <[EMAIL PROTECTED]>
To:
Sent: Monday, September 18, 2006 8:34 PM
Subject: [asterisk-users] Fedora
> Hi list;
>
> Does asterisk work with fedora because redhat
>
bilal ghayyad wrote:
Hi list;
Does asterisk work with fedora because redhat
enterprise is licensed and costly.
If you want enterprise class software look at CentOS. http://www.centos.org
But in answer to your question, Fedora should be supported. There are
some quirks with Redhat distros
Yes. We use fedora as a test system before moving it onto RHEL.
On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote:
> Hi list;
>
> Does asterisk work with fedora because redhat
> enterprise is licensed and costly.
>
> Regards
> Bilal
>
> __
>
We use asterisk on Fedora Core 3 without any issues in one of our
installations.
If you're after Redhat without the cost, though, you might check out CentOS as
a distribution. :)
Fedora can sometimes be a little bleeding edge when it comes to a production
environment.
N.
On Mon, 18 Sep 2006
Hi list;
Does asterisk work with fedora because redhat
enterprise is licensed and costly.
Regards
Bilal
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Hi
I just want to note some technical points here:
On Mon, Sep 18, 2006 at 12:08:20PM +0100, Nick Burch wrote:
> On Mon, 18 Sep 2006, Klaus Darilion wrote:
> >Does some one of you have practical experience with the Astribank
> >channel bank (I've tried to contact xorcom directly but didn't recei
I'm trying to make
the Polycom 501 go off-hook (in speaker phone mode) when any digits is dialed
and the handset hasnt been lifted. Is this
possible?
Mike
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On Sun, Sep 17, 2006 at 03:34:05PM +0200, Paul Hewlett wrote:
> On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote:
> > On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote:
> > > I'm banging my head on compiling bristuff modules for Suse 10.0 with
> > > kernel
> > >
> > >
> > > Linu
Hello list. I have Asterisk installed on my small home Linux machine, and it was
working with softphones. I later got a X100P card (I think it was a clone, buit
Asterisk saw it as a real X100P). I then got side-tracked and the hard drive
later failed in the machine. I've replace the whole machine,
On Monday 18 September 2006 04:53, Giorgio Incantalupo wrote:
> What I do is:
> 1) use "chanisavail" command to ask Asterisk for a free channel to use
> 2) use that channel to dial outbound calls
The problem is that a race condition exists; the channel could be available at
time (1), but somethin
I've always had ignorepat => 9 followed by _9. in my outgoing context
now if I use this in 1.4 it jumps straight to BUSY.
I also have _70. and _80. which make calls through other * servers, they
work as expected.
What is really surprising is that _90. also jumps straight to BUSY.
Any ideas?
--
On Mon, 2006-09-18 at 10:53 +0200, Giorgio Incantalupo wrote:
> Hi,
> I have an Asterisk box on a Debian Sarge distro with a Sangoma a102 PRI
> card and I have a big problem: Asterisk drops a lot of outbound calls.
> I do not use groups, I want to use a free channel given from Asterisk.
>
> What
It's up on http://www.freedomphones.net/polycom/files/ now with release notes.
MATT---
On 9/18/06, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
Hi, all -- since the new 2.x firmware seems to support NAT -- and, since
I'm not an authorized dealer -- I'm kind of wondering if anyone knows
where I ca
Hi All
Anyone have any knowledge of using the above?
I have installed it and wish to send out a fax via a SIP channel on an
ISDN3O. I have used the test script which says it has gone though ok but
i never see any activity in asterisk.
Checking the /var/spool/asterfax/tmp directory i can see a
On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote:
> On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote:
> > I'm banging my head on compiling bristuff modules for Suse 10.0 with
> > kernel
> >
> >
> > Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64
> > x86_
Hello,
have a look here .
ftp://nxs.yi.org
Harry
--- Ken D'Ambrosio <[EMAIL PROTECTED]> a écrit :
> Hi, all -- since the new 2.x firmware seems to
> support NAT -- and, since
> I'm not an authorized dealer -- I'm kind of
> wondering if anyone knows
> where I can get it.
> freedomphones.net/pol
Hello, Does anyone have a solution for having SIP users to authenticate against a LDAP server?Best Regards,Andre O.
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Im going to get a trial account, .014 to US is not bad at all!Only downside is that g729 is only codec they allow :(On 9/17/06,
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I saw this termination company, www.BuyMin.com the rates looks good. Has anyone any experience with this company? I use Gafa
On Mon, Sep 18, 2006 at 08:22:03AM -0700, Nick Ellson wrote:
>
>
> Thanks for the feedback Peter,
>
> I am going to try one with an FXO and then one of the $30 fixed port
> single FXO PCI cards from pbxeq.com as well. See if there is a real
> difference there. Anybody try the "A-100PCI" card?
Hi,
we have experienced som troubles with the timeout option of the
Dial-App. It seems the Dial startts counting down the timeout imediatly,
but there are great differences when the called phone actually starts
ringing. If i call a landline phone in my own country it is nearly the
same, but if i w
If you have a PRI, why are you telling asterisk which channel to use?
why don't you set groups in zapata.conf and just use g1 or the like?
On 9/18/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:
Hi,
I have an Asterisk box on a Debian Sarge distro with a Sangoma a102 PRI
card and I have a big
Remi Quezada wrote:
Hi,
Right now I am in the process of setting up an asterisk box. I was
thinking of having two asterisk box, one that is hooked up to the PSTN
using a digium TE405P card and the other asterisk box will be used to
store all the sip user features and routing information. Do yo
Keep in mind that the Mediatrix does not support register (AFAIK,
anyhow). You have to create a static entry in sip.conf that has host
set to the IP address of the Mediatrix
On 9/18/06, Bill Michaelson <[EMAIL PROTECTED]> wrote:
Thank you, C F and Florian. Now I must expose my ignorance about SI
I'm testing it against the Aastra 9133i atm. I am trying to rely on the 9133i native BLA/SLA over Asterisk. The hard part is trying to figure out how to do it atm.MikeOn 9/18/06,
Olivier <[EMAIL PROTECTED]> wrote:
2006/9/18, mike pham <[EMAIL PROTECTED]>:
3) with * version 1.4 you can configure
Thanks for the feedback Peter,
I am going to try one with an FXO and then one of the $30 fixed port
single FXO PCI cards from pbxeq.com as well. See if there is a real
difference there. Anybody try the "A-100PCI" card? When I do, I'll post
what I find.
Nick
--
Nick Ellson
CCDA, CCNP, CCS
I know this isn't directly Asterisk related. - But I do appreciate the
responses I get from you folks. - At least I don't get flamed like I've seen
on the Java & Perl-Mod lists (geesh!).
Okay - Here's what I've got.
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll
work
Here is the log excerpt from rright before the crash:
Sep 18 09:58:58 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403415-0a2d8020
Sep 18 09:59:17 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403425-0a2f0cf8
Sep 18 09:59:53 WARNING[23698] pbx.c:
Do you have a backtrace so we can see where it crashed and have you reported a bug with the backtrace?Joshua ColpSoftware DeveloperDigium, Inc.- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, September 18, 2006 11:45:14 AM GMT-0800Subject: [asteris
Hi, all -- since the new 2.x firmware seems to support NAT -- and, since
I'm not an authorized dealer -- I'm kind of wondering if anyone knows
where I can get it. freedomphones.net/polycom/files/ only goes up to
1.6.7. If anyone can either mail it to me, or mail me a link, I'd
certainly be apprec
Nick,
I use one and it works just fine for me with 2 FXO and 2 FXS at the
moment. I would say it is a great board to have and experiment with and
as you say not too big or too small.
Peter
Nick Ellson wrote:
I know it's not a digium product, but the 12 port A1200P card with a
single FXO m
Yes, I deleted the old modules. All the modules come up listed as the date of the upgrade.
Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection.
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Hi all,
the default message for email notification looks like:
New 0:09 long msg in box 2001
from XliteUser2002, on Monday, September 18, 2006 at
04:24:11 PM
i try to change it with "emailbody=" but i always get
the default message body.
my voicemail.conf looks like
[general]
format=wav49|gsm
Hi all,
> Frank,
> this were my first thougts, why the hell does he say
> something like this
> on asterisk and doues not say whats wrong!
I've spent some time to have both asterisk and ser
working together.
In my last post I asked how asterisk could send invite
to a outboundproxy (SER) becaus
I had FOP installed previously on [EMAIL PROTECTED] which was working perfectly fine until today when I decided to upgrade it to its ver. 2.6
I ended up losing my old FOP and new one no success.
First I backed up /var/www/html/panel to /var/www/html/panel_old. But after no sucess with new FOP,
Thank you, C F and Florian. Now I must expose my ignorance about SIP and
Mediatrix...
I've adapted my sip.conf to essentially conform with what you've posted.
So when I restart the Asterisk server, ethereal indicates that a NOTIFY
goes to the Mediatrix (at 192.168.20.188), which responds with
Title: Re: [asterisk-users] Chanspy crashing the server, again
Anybody have a backtrace?
On 9/18/06 9:23 AM, "Richard" <[EMAIL PROTECTED]> wrote:
I am experiencing the same problem.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, September 18,
I am experiencing the same
problem.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Monday, September 18, 2006 10:03
AMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users] Chanspy crashing the server, again
I upgraded to 1.2.12.1 - the prob
Are you sure you deleted all the old asterisk modules when upgrading?
On 9/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was
once or twice per week, now EVERY TIME someone uses chanspy it crashes the
machine.
Anyone wi
I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine.
Anyone with any ideas? This is killing me.
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what about this?
show app ringing?
[incoming]
exten => s,1,Answer
exten => s,n,ResponseTimeout(5)
exten => s,n,Playback(mymessage,skip)
exten => s,n,Background(mymessage2)
exten => s,n,Background(silence/3)
exten => _7XX,1,Ringing
exten => _7XX,2,Goto(local,${EXTEN},1)
[local]
ext
go here
http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate
and read ;)
Siqhamo Sifo schrieb:
How does one configure user authentication on asterisk .
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T
Hi,
Right now I am in the process of setting up an asterisk box. I was
thinking of having two asterisk box, one that is hooked up to the PSTN
using a digium TE405P card and the other asterisk box will be used to
store all the sip user features and routing information. Do you think
this a good de
How does one configure user authentication on asterisk .
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2006/9/18, mike pham <[EMAIL PROTECTED]>:
3) with * version 1.4 you can configure 100 and 111 as a sla group and pickup the call at either location using the hold button
Hi Mike,Have you tried yourself SLA already ?Would you advise someone to use it on a production server now ?Regards
_
I have auto attendant menu. When calling person dials one number one extension
rings. Problem is that while extension rings caller doesn't hear ringing. I
understand that caller doesn't hear ringing because phone call is already
established, but I need to "tell" to caller that extension is ringi
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