Asterisk has an option to have an out (by pressing '0') and you could
use that to jump out of voicemail and off to someones mobile.
Maybe a dbget to grab the mobile phone for the user would be a neat way
to go.
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
ph: 03 8320 810
Currently we have (with our NEC phone system) the options in voicemail to
have a message say " press 2 to go to my mobile phone"
Can this be done in asterisk without setting up an IVR for each user ?
Has anyone got a voicemail dialplan that can do this ?
Thanks
--
Kevin Withnall
ILB Computing
Hi Michael,
I have a TDM400P on an Asterisk box with:
1) a FXO connected to the old pbx and
2) a FXS connected to a normal analog phone
3) the analog phone is a Telecom Sirio, (the most common in Italy)
If I knew how to check asterisk send/receive this non-digits signals it
can be easier to u
Hi Dmitry,
thank you SO much for the help. No one else seems to be stuck with the SQL Server, Unfortunately you and I are. Don't know your reasons but I had no choice but to use that. Anyway, the inability to see the "view" was related to a cross-permissions issue as the vmuser view is being crea
Hello, RR!
Yes, that is all that you need to fix the freeTDS
"pseudo-thread-safe" problem.
And you should know nothing else to enjoyable using of Asterisk!
Congratulations!!! :)
I spent much more time to find the problem, find and apply the
appropriate patch, fix some bugs, then apply that
Greetings,
This weekend I had some free time, and decided I would try and get the
Hudlite Call Manager working with my Asterisk Installation. This
wasn't the easiest of processes since I do not run TrixBox, FreePBX,
etc. I have a stock Asterisk installation on Gentoo Linux, managed
from command l
If you are using putty as your ssh client... create the ssh tunnel to the * box & then go to "session-> logging-> log printable output only" in your putty configuration & save.
To do it on the asterisk box only, I only know of the logger:
1. In logger.conf set "full => warning,error,verbose,deb
Herchi,I want you to re-read my last e-mail very carefully. Your response does not mention at all my guess that the three SP_DIRSRVR variables may be giving you trouble. I'm still interested in knowing what happens if you remove them from your settings file.
Also, I have heard a rumour that there
Well guys.
Another day of shutdowns.
[Jul 3 07:00:08] VERBOSE[3086]: [Jul 3 07:00:08] Beginning asterisk
shutdown
[Jul 3 07:02:23] VERBOSE[2657]: [Jul 3 07:02:23] Beginning asterisk
shutdown
[Jul 3 10:59:01] VERBOSE[3083]: [Jul 3 10:59:01] Beginning asterisk
shutdown
[Jul 3 11:
Hello world,
Any success stories of getting a Nokia E61 to work with Asterisk
server? Interested to hear before we buy them for work :)
Thanks for your time
___
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Asterisk-Users mailing list
To UNSU
On Mon, Jul 03, 2006 at 07:33:31PM -0400, William F. Acker WB2FLW
+1-303-722-7209 wrote:
> On Mon, 3 Jul 2006, Tzafrir Cohen wrote:
>
> >On Mon, Jul 03, 2006 at 02:04:43PM -0400, William F. Acker WB2FLW
> >+1-303-722-7209 wrote:
> >>Hi all,
> >>
> >> Since I'm restricted to the text console,
On Mon, 3 Jul 2006, Tzafrir Cohen wrote:
On Mon, Jul 03, 2006 at 02:04:43PM -0400, William F. Acker WB2FLW
+1-303-722-7209 wrote:
Hi all,
Since I'm restricted to the text console, at least for the near
future, I use Asterisk as a softphone. My new machine uses the new family
of Intel so
Salve C, salve Tzafirir!
What a short name
On Mon, 03 Jul 2006, C F wrote:
> Thanks Tzafrir, I realy enjoyed reading it.
me, too.
> On 7/3/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> >[off-topic]
Not 100%, I'm a asterisk newbee and think that the
rigth editor with syntaxhighlighting can
This has been my experience as well. I also posted the issue to this
mailing list, but has not responses. I have not come up with a
workaround. If I have time I'll try to write up a bug report, but it
will be a while. You are welcome to document the issue with as much
detail as you can and
Also, This is the information that displays on the console:
-- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX, requested format =
4, actual format = 4
-- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]/6", "XXX-XXX-|a") in new
stack
-- Executing SetCIDName("IAX2/[EMAIL PROTECTED]/6", ""SOME COM
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten => _9011.,1,SetCIDNum(XXX-XXX-|a)
exten => _9011.,2,Set
On Mon, Jul 03, 2006 at 02:04:43PM -0400, William F. Acker WB2FLW
+1-303-722-7209 wrote:
> Hi all,
>
> Since I'm restricted to the text console, at least for the near
> future, I use Asterisk as a softphone. My new machine uses the new family
> of Intel sound chips. I imagine that I need
On Mon, Jul 03, 2006 at 08:12:40AM -0700, chawki hammoud wrote:
> Hi:
>
> I have three TDMs that were working fine on my
> asterisk system. Now I receive an error when I try to
> modprobe wcfxs. Something I didn't have before:
Which version of zaptel do you try to install?
wcfxs of zaptel <= 1.0
I have a feeling that going after the C code will be easier than using a
multiple dial to create a conference, join it, play the file and all such
things. It would be very nice, in any case, if the queue app had dialplan
callback points, so that you could make such things easier
l.
I
glad to help...
i was actually writing a response about how i didn't know why yours
wasn't working when it hit me - in a macro, you can't have anything
other than the s extension defined. in other words, you can't have
1,1 or 2,1 - any keypress breaks the macro.
in fact, that is EXACTLY why i d
Hi all,
Since I'm restricted to the text console, at least for the near
future, I use Asterisk as a softphone. My new machine uses the new family
of Intel sound chips. I imagine that I need to specify a device in
alsa.conf, but I haven't a clue what this card requires.
Any ideas?
Hello all,
Quick question... I have the Zap dev kit. When ever I try to make a call from
Zap/1 (FXS) via Zap/4 (FXO) I can hear the person on the call fine but they
complain that I am muffled. this problem is not there if I call from one of my
pap2's
Also the call is fine if made from the PA
Thanks for the response! I used your template to write a similar one for us
and it
works great. I wonder if there is a bug in the macro timeout code.
David
whois wes said:
> This may sound stupid, but I had a similar issue that I solved by
> placing an Answer at the beginning of what would be
The MGCP versions will have a "-CS" annotated at the end of the part number.
Ex IP500CS
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message -
From: "Eric "ManxPower
After ring I hangup phone but don't speak at the phone silence :-
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Monday, July 03, 2006 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
There are different versionsof the polycom phones. Depending on the
actual part number it can come with MCGP, SIP, or H323. Polycom does
not support customer migration from one protocol to another. Get the
version with the SIP firmware.
Jim Freeze wrote:
Hi
I was about to order a polycom
The dial macro works only for the agent called and not for the caller...
I already use it and it's quite usefull to play different annoucements
to the agent but that's useless for the caller as the bridge is done
after the end of the macro...
I only need to tell the caller that his line has b
Hi Chawki.
Make a test, remove the symbol "wcfxs.ko" in directory /lib/modules/2.6.8.1-12mdk/misc/ and recompile zaptel, libpri and asterisk.
Good job!
Regards
Josué
2006/7/3, chawki hammoud <[EMAIL PROTECTED]>:
Hi:I have three TDMs that were working fine on myasterisk system. Now I receive
On Mon, Jul 03, 2006 at 06:16:11AM -1000, Jean-Denis Girard wrote:
> Dinesh Nair a écrit :
> >
> >cant seem to get it to work on Mozilla/5.0 (X11; U; FreeBSD i386; en-US;
> >rv:1.8) Gecko/20051228 Firefox/1.5. any chance this is on the radar ?
> >
>
> Hi Dinesh,
>
> It should be working. What ha
Use features.conf,
look here at the comments:
http://www.voip-info.org/wiki-Asterisk+cmd+flash
On 7/3/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:
Hi C F,
you say Flash asterisk command send a flash signal to old pbx so that it
sees that command as coming from an analog phone. But since Fl
Dinesh Nair a écrit :
cant seem to get it to work on Mozilla/5.0 (X11; U; FreeBSD i386; en-US;
rv:1.8) Gecko/20051228 Firefox/1.5. any chance this is on the radar ?
Hi Dinesh,
It should be working. What happens exactly: is this an installation
problem, or what ? Can you try running Firefo
Perhaps the BT crew are all on a drunken rampage along Sochiehall
Street?
On Mon, 2006-07-03 at 15:14 +0100, Colin MacMillan wrote:
> Hello,
>
> For some reason I can't call Scotland from London ...
>
> The details:
> Asterisk v. 1.2.9.1
> ISDN2 Interface - Junghanns card with BRIstuff 0.3.0-P
Hi Dmitry,
just to answer your questions, and telling you what I've done so far,
1) yes, using FreeTDS
2) yes, configured unixODBC with FreeTDS to talk to MSSQL. I have all of this working where I can load and update tables in MSSQL using the "realtime load/update" statements from the Asterisk
http://svn.digium.com/view/asterisk/trunk/UPGRADE.txt?view=markup
* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
created channel. This variables holds the channel name of the transferer.
I'm not sure if channel name is enough for me, but i made the the cid_num
Hello again
I have looked it up, it is the latest software. But they have released a new
minor update. According to changelog only added support for new operators.
I flashed a E61 with new software (Same kind of minor update) and it did the
trick for it. Couldn't get voip to work before, but wor
TDMoE doesn't seem to be a good alternative.
it doesn't make sense to use an eth-interface used for
intranet-traffic/sip/sccp as well
...to heavy load to get a reliable function. On my test-asterisk with just
activated ztd_eth-module and configured zaptel it filled up my log with
error-messages
> you say Flash asterisk command send a flash signal to old pbx so that
it
> sees that command as coming from an analog phone. But since Flash is
not
> a digit, how can I catch it from within asterisk? How can I tell
> asterisk (es inside extensions.conf) to do something whene receive it
> from a p
I'm not sure you can do it if you want a play a file to the caller when
your agents pick up the line. I believe you may want to use a dial macro -
option M() - in the dial command that the local channel runs in order to
connect to the callback agent. see
http://www.voip-info.org/wiki-Aste
OK so * is seeing the phone go offhook That is good
How about if you call the handset, can you actually talk across the
connection? You said it rang, but now we need to establish you
actually have an audio connection.
On Jul 3, 2006, at 9:50 AM, Viktor Tatianin wrote:
If lift up handset
You can get the same result - well, maybe better - using a monitorin tool
that will display the availabe calls in the queue. QueueMetrics does it,
and FOP will do this too.
Hope this helps
l.
On Mon, 03 Jul 2006 15:09:44 +0200, Mailing Lists
<[EMAIL PROTECTED]> wrote:
Is it possible
Hi C F,
you say Flash asterisk command send a flash signal to old pbx so that it
sees that command as coming from an analog phone. But since Flash is not
a digit, how can I catch it from within asterisk? How can I tell
asterisk (es inside extensions.conf) to do something whene receive it
from
Show channels
asterisk1*CLI> show channels
Channel Location State Application(Data)
Zap/94-1 [EMAIL PROTECTED]:1Rsrvd (None)
1 active channel
0 active calls
But at phone is silence
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROT
Hi:
I have three TDMs that were working fine on my
asterisk system. Now I receive an error when I try to
modprobe wcfxs. Something I didn't have before:
modprobe wcfxs
FATAL: Error inserting wcfxs
(/lib/modules/2.6.8.1-12mdk/misc/wcfxs.ko): Unknown
symbol in module, or unknown parameter (see dme
Thanks a lot, Fabio, I've been looking for this page.
Zerthimon.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fabio
Sent: Monday, July 03, 2006 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with
Hi GroupThanks to Alexander and Julio for the supportWhat Julio says is truth, I want to use the e-MTA SBV5120 from Motorola and to use * as CMS. And PacketCable VoIP use NCS but no SIP.Alexander has good conclusion but for what I read, Asterisk had a proyect apart to work with PacketCable which wa
This may sound stupid, but I had a similar issue that I solved by
placing an Answer at the beginning of what would be your 'screen'
macro.
However, we have a call-screening macro as well, and don't have the
Answer command at the beginning...here it is, if this helps any.
[macro-announce]
exten =
If lift up handset
3 17:48:57 VERBOSE[14197] logger.c: -- Starting simple switch on
'Zap/94-1'
Jul 3 17:49:14 DEBUG[14197] chan_zap.c: not enough digits (and no ambiguous
match)...
Jul 3 17:49:20 DEBUG[14192] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jul 3 17:49:21 DEBUG[14197
Title: Message
You
can run MGCP on Asterisk, but you should be able to find a 301 with
SIP. If not, try another vendor.
-Nate
-Original Message-From: Jim Freeze
[mailto:[EMAIL PROTECTED] Sent: Monday, July 03, 2006 9:12
AMTo: Asterisk Users Mailing List - Non-Commercial
Agreed, but it would be a real shame to have to deal specially with
Aastra phones when disabling CWI should be a feature of the phone!
:-( Apart from this, they are really nice (and cheap) phones.
Steve
On 7/3/06, Jerry Jones <[EMAIL PROTECTED]> wrote:
You can always control via the dialplan f
To add to the mystery, if the cell phone answers and presses "1" as requested,
the
logs don't register priority 1,1 being executed. It is as if the macro has
prematurely aborted.
David
David said:
> I just downloaded, compiled and installed Asterisk 1.2.9.1. I did this
> specifically
> to get
MGCP stands for Media Gateway Control
Protocol. You want the SIP versions of the 301 and 501, both of which
integrate nicely with Asterisk.
Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY 14225++voice - 716.630.1555
X22email - [EMAIL PROTECTED]AIM - B2CORY
I don't realy get it, what are you trying to accomplish? that the CID
should be that of who?
On 7/3/06, Kai Ober <[EMAIL PROTECTED]> wrote:
thx, but where is the CALLERID in ${BLINDTRANSFER}, ther is just
cheannel info.
looking for this // , not this
/
> http://www.voip-info.org/wiki/view/BLIN
Thanks Tzafrir, I realy enjoyed reading it.
On 7/3/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
[off-topic]
On Sun, Jul 02, 2006 at 10:56:03AM -0400, C F wrote:
> I believe vi is GPLed
Thanks for asking.
The original vi was part of BSD, I believe. I'm not sure what it's
license was.
nvi is u
The flash command will do just that. However flash only works on FXO
ports and not on SIP FXO ATAs, if you use the later then you will have
to find out how your ATA supports it.
The easiest way to set this up is to use the features.conf
On 7/3/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:
Zerthimon,
we have something very similar, and i call it our 'agent IVR'
when an extension is run and nobody answers or the phone is
unreachable, the caller is given options to continue holding, be
connected to customer service, or go to voicemail. similar to what
you want to do. you'd need to
Are you seeing any messages on the console? You should be seeing
something like "Starting simple switch"
We would need more info to help more.
On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote:
Hello All
Please help me, I have next problem
When lift up handset at phone which connect to ch
Hello,For some reason I can't call Scotland from London ...The details:Asterisk v. 1.2.9.1ISDN2 Interface - Junghanns card with BRIstuff 0.3.0-PRE-1qExtensions.conf
(context SIP-PHONES)exten=>_X.,1,Dial(Zap/g1/${EXTEN},60)When I call this number - 01417778979 (this is a building company and the nu
Two's enough for me! I'll open a bug.
Doug.
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Mon 7/3/2006 4:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] As
HiI was about to order a polycom 301 when I noticed that the VoIP protocol is listedas MGCP and not SIP, as with the 501.First, what is MGCP?And, will the 301's work seamlessly with Asterisk and the other 501 phones that I have?Thanks Jim Freeze ___
--Ban
You can always control via the dialplan from *
Just starting to play with the Aastra so not much knowledge on them yet.
On Jul 3, 2006, at 6:41 AM, Steve Davies wrote:
Hi,
Does anybody know whether it is possible to completely disable call
waiting on an Aastra 9112i?
The latest 1.4.x firmwa
Hi,
should I care about this error message in /var/log/asterisk/message?
Jul 3 15:18:42 WARNING[31782] file.c: Unexpected control subclass '14'
Jul 3 15:19:38 WARNING[31792] file.c: Unexpected control subclass '14'
Jul 3 15:21:08 WARNING[31837] file.c: Unexpected control subclass '14'
Jul 3 1
Khaled Chehab wrote / skrev:
Is SRTP available in asterisk?
In a SVN branch.
See http://bugs.digium.com/view.php?id=5413
Mikael
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To UNSUBSCRIBE or update op
On Wed, 2006-08-02 at 16:22 +0300, Khaled Chehab wrote:
>
>
> Is SRTP available in asterisk? Or how to implement it ? am using
> trixbox
1) fix the time on your PC
2) all questions regarding trixbox should go to the trixbox mailinglist
or forum
3) make it a habit of asking google first
4) s
Most likely, he is thinking something like using the MTA (a motorola
cable modem with RJ11 phone ports), to register to Asterisk.
From what I understand, most (if not all) packet cable VOIP is done
using NCS (a mgcp-like protocol ?) as call control, not SIP.
Alexander Lopez wrote:
Isn’t DOCSI
Hi,
I have to connect an old PBX to a new Asterisk box. but I must keep the
same flash button functionality of the old system. Is it possible to
tell asterisk to send a Flash signal to old pbx when receiving it from a
phone? I know there is a flash command inside asteriskis there
anybody
Is SRTP
available in asterisk? Or how to
implement it ? am using trixbox
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written
Hello All
Please help me, I have next problem
When lift up handset at phone which connect to channel bank I don't hear
dialtone but if I dial this number phone ring
When after ring hungup handset at phone voice not work
Thanks
Viktor Tatianin
___
--
Wouldn't it be nice if this fellow fixed his time and date?
John Novack
Khaled Chehab wrote:
I added a context and assigned on it an international number to be
dialed from the asterisk directory *411 as shown below
But the call doest succeed (declined) how can I let directory use my
Is it possible for an agent using agentcallbacklogin to have
callwaiting? If the agent already has a call and another call is queued it
would be nice if the agent could see that there is more calls in the queue...
/urban
___
--Bandwidth
See bug 6315:
http://bugs.digium.com/view.php?id=6315
I think that this was MEANT for inclusion into the source but never
quite made it, except in the documentation. I have been meaning to
request that the bug be reopened, because we rely heavily on inbound
call queues and need that functionali
Hi Zerthimon
take a look at http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
regards
Fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Lior Goikhburg
Enviado el: Lunes, 03 de Julio de 2006 07:48 a.m.
Para: asterisk-users@lists.digium.com
Asunto
hi list,
i have tried to set the call waiting function using freePBX but it dosent
work. i think there is something wrong with the coding. Has anyone
experienced this sort of problems?
Can you expand a bit more on your problem? What versions of software are
you running? What have you tried
Hello Robert,
You need to have include files, try compile and install asterisk
before addons ;-)
Monday, July 3, 2006, 6:36:39 AM, you wrote:
> build_tools/mkdep -fPIC -fPIC app_addon_sql_mysql.c app_saycountpl.c
> cdr_addon_mysql.c res_config_mysql.c
> app_addon_sql_mysql.c:15:22: error: as
Hi,
I had edited out all lines starting with a #, which is ot
right, as the marker for comments is ##... See below for the entire
file.
I just tried the configuration through DHCP, by
setting the 176 option to point to the right TFTP server and also to the right
SIP proxy. The Avaya boot
I added a context and assigned on it an international
number to be dialed from the asterisk directory *411 as shown below
But the call doest succeed (declined) how can I let directory
use my outbound rule
[ck2]
9613504768 => 111,kiki,,,attach=no|saycid=no|envelope=no|delete=
Hi,
Does anybody know whether it is possible to completely disable call
waiting on an Aastra 9112i?
The latest 1.4.x firmware allows the call waiting /tone/ to be
disabled, but I have users who really do not want a second call to
ring them at-all if they are busy, yet the Aastra seems to insits
Ok, you use the MS SQL. It's problem of FreeTDS implementation details.
1. Did you install FreeTDS or other ODBC driver to MS SQL(Sybase)
connecting?
2. Did you configure unixODBC to use freeTDS driver (etc\odbc.ini,
etc\odbcinst.ini)?
3. APPLY the patch that I tell you before
( http://bugs.di
Mike,
If you feel afraid of the next power outage, why not install a more
powerfull UPS with a longer run time? Or, as it is in my case, a friend
of mine substituted the factory default battery in the UPS with a car
battery, that holds the Server for 4-5 hours. Add another battery and it
will
Hi Dinesh,
Thanks for your suggestion - however, just for testing it out, are there
any softphones out there that support paging/intercom? Would auto answer do?
Regards,
Mike
Dinesh Nair wrote:
On 06/29/06 01:18 Jeremy McNamara said the following:
why not setup a listen only meetme for th
Dear
I am using trixbox,I want ot disable and enable voicemail
from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was
working successfully
Database put AMPUSER/9990999 voicemail default
And
Database put AMPUSER.9990999 voicemail disables
Bu
Bryan Field-Elliot wrote:
For the typical home user who has a SIP ATA behind (usually) a Linksys
home router/firewall, what's the best practice "qualify=" time we should
be running on the server, to keep the home user's NAT happy?
The default, 2 seconds, is way too short (generates too much ne
Hi everybody !
I need to play a file to customers when an agent answered the line to
tell them it's their turn but I don't want to do periodic annoucements,
Is there a way or something I misunderstood in the voip.org docs because
I can't do this for the moment !
Thanks,
Tristan
__
Marcin Lukasik wrote:
Have you even _tried_ to create your "dialplan"?
And to make it worse, he copied this drivel to the Developers lists.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTE
I have definitely run into this on the one production site I have with
1.2.9.1
I haven't tried backrevving in order to see if it affects 1.2.9 or
older, but it is very annoying.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sen
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Hi Dmitry
Thanks for that. Yes, I had done all of that and have a working Asterisk <-> MS
SQL setup where I have a view for the voicemail.conf settings in my database +
I have a separate table called 'voicemessages' to store my voicemails. What I
wanted to know was that if I have got this far w
Greetings,
I'm
an asterisk newbie. I have no problem making "flat IVR trees" but
this one is too confusing.
I'm
trying to set up a dial plan with a multi-layered IVR:
I've
been asked to create a menu for my company where one could dial a menu entry
like: "for sales press 1, for man
Oh! I'm sorry. :*)
You are right - it's about freeTDS problem. Just for case of the MS SQL
storage usage.
You need to configure the Asterisk and unixODBC. Read, please next
listed articles:
http://www.asteriskguru.com/tutorials/realtime_pgsql.html (it's for
Postgre SQL. You need use drive
The following page will give you a hint:
http://nerdvittles.com/index.php?p=73
Regards,
Tigran
Hello Sharon,
Saturday, July 1, 2006, 5:33:14 AM, you wrote:
> anyone have information on how the call back features work with
> asterisk? I means the dial plan or what so ever. thanks
--
Best re
On 07/03/06 15:41 Jean-Denis Girard said the following:
MozPhone no longer depends on any external libraries (libiaxclient is
statically compiled in, and jslib is now included). So install is very
simple, like any other firefox extension. It is correct that newer
cant seem to get it to work
Hi Dmitry,
Thanks so much for replying, I had a read of the bug descriptions you pointed
to but unfortunately am not a programmer so have a bit of a struggle following
the entire discussion. I actually AM using 1.2.9.1. So does that mean that I
should be OK and do not need any patches as this r
Dear
I am using trixbox,I want ot disable and enable voicemail
from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was
working successfully
Database put AMPUSER/9990999 voicemail default
And
Database put AMPUSER.9990999 voicemail disables
But a
[off-topic]
On Sun, Jul 02, 2006 at 10:56:03AM -0400, C F wrote:
> I believe vi is GPLed
Thanks for asking.
The original vi was part of BSD, I believe. I'm not sure what it's
license was.
nvi is under a BSD license.
levee seems to be distributed with a MIT license of some sort.
elvis is distr
On 3 Jul 2006, at 07:52, Dinesh Nair wrote:
On 07/03/06 12:51 Tzafrir Cohen said the following:
Web pages, evenwith javascript, are still very limited. For instance,
they cannot establish UDP communication on their own with other
places.
An arbitrary TCP connection is also not so trivial.
hi list,
i have tried to set the call waiting function using freePBX but it dosent work. i think there is something wrong with the coding. Has anyone experienced this sort of problems?
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Hi, RR:
Asterisk 1.2.1 with listed in
http://bugs.digium.com/bug_view_page.php?bug_id=5756
patch have some problems with voicemail storage, but that problems
already fixed in the more newer version.
I'll advice you to try using 1.2.9.1 version with the same patch that
I'd applied to it (see ht
On 3 Jul 2006, at 09:57, Amund Nygaard wrote:
As far as i know there are only support sites for the service
center. I can try and look it up. Mine has 1.0610.02.15
Mine has the same version. If you could double-check that this is the
latest version that would be great!
jens
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On Mon, Jul 03, 2006 at 02:52:52PM +0800, Dinesh Nair wrote:
>
> On 07/03/06 12:51 Tzafrir Cohen said the following:
> >Web pages, evenwith javascript, are still very limited. For instance,
> >they cannot establish UDP communication on their own with other places.
> >An arbitrary TCP connection is
As far as i know there are only support sites for the service center. I can try
and look it up. Mine has 1.0610.02.15
I just started using a brand new one, also lost connection during the weekend,
and reboot needed.
According to Nokia there will come a phone called 6136 soon, that will allow
h
On 7/3/06, Jean-Denis Girard <[EMAIL PROTECTED]> wrote:
But I'm not sure that MozPhone is what the original poster asked.No, however, I like to read all these different point of view.
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