Use the macro feature in dial (CVS-HEAD only, or apply the patch)
documented here:
http://www.voip-info.org/wiki-asterisk+cmd+dial
On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote:
Dear All,
I like to implement something that does the following:
- a call comes in
- answered: Please
At 04:19 PM 17/04/2005, C F wrote:
Use the macro feature in dial (CVS-HEAD only, or apply the patch)
documented here:
I can't see a way to get Queue to use the macro; it has a limited number of
options available.
I have tried using 'Local/[EMAIL PROTECTED]' as a queue member, but this seems to
Don't use it with queuing, use it with dial
On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote:
At 04:19 PM 17/04/2005, C F wrote:
Use the macro feature in dial (CVS-HEAD only, or apply the patch)
documented here:
I can't see a way to get Queue to use the macro; it has a limited number of
I'm posting this here because I'm betting many of you are qualified
and someone may be interested. Please, no flames, just act if this is
something that interests you, it may be worthwhile. If not move on.
Saw this on comp.dcom.voice-over-ip. I want and looked at the site and
they do have
I still cannot find it:
What does it mean, and how can I fix it?
Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
Any idea?
-- SIP Seeding peers from Astdb: '3366' at
[EMAIL PROTECTED]:64440 for 3600
-- Saved useragent Sipcom/ATA2000-1.6.11 for peer 3366
-- SIP Seeding peers from Astdb: '886229421761' at
[EMAIL PROTECTED]:5060 for 3600
-- Saved useragent Grandstream BT100 1.0.5.18 for
I tried many different possible ways to us speed dialing, however, I
end up in the default context, where the number does not match anything,
... with the result Playing 'demo-congrats'
I also could not figure out how to use the tabs Queues and Agents
I have not found a new version over the
Hello everybody,
I recently got a new phone line from Bulldog-CW (UK). Needless to say
that I have connected phone line to my Asterisk system. All seems ok but it
does not detect hangups. When the caller hangs-up, the Bulldog line gives a
continues tone for a few seconds and then it goes
Hi:
I want to use G729 codec from my iax connection to my
voip provider and later between my two asterisk boxes.
G729 bandwidth requirement is relatively low and I
intend to reduce more by applying point-to-point-link.
While trying to do my homework and figure out how to
do that, I appreciate
Hi,
I managed to set up two CISCO 7940 phone with a SIP firmware 7.0, with
Asterisk. At the beginning though, I coulnd't understand why they
wouldn't work, even if I followed all the instructions found on
voip-info.org.
Eventually, after some debug and with someone else help, I managed to
make
Hi Ronald,
You posted he same question yesterday and I answered you. Do you till have
problems?
Thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
I tried many different possible ways to us speed dialing, however, I
end up in the default context, where
Hi,
I managed to set up two CISCO 7940 phone with a SIP firmware 7.0, with
Asterisk. At the beginning though, I coulnd't understand why they
wouldn't work, even if I followed all the instructions found on
voip-info.org.
Eventually, after some debug and with someone else help, I managed to
make
Hi
Grabbed the most recent stable asterisk from CVS as documented here:
http://www.asterisk.org/index.php?menu=download
Didn't bother with zaptel or libpri as I have no Digium hardware nor T1 or
E1.
Did
make install asterisk; make samples.
Started asterisk with
asterisk -c
and it
tgj wrote:
Hi Ronald,
You posted he same question yesterday and I answered you. Do you till have
problems?
Thank you for posting yesterday that there is a new version available,
I still have the same problem, I cannot get it to work
Thank you that your reply now includes a working
Version 0.91 - 17. April 2005.
* IPS is now using the context configured in Asterisk for peers - the
context on the configuration page is used for Speed Dial Numbers only
* New tab page for Speed Dial Numbers
Download here: http://ipswitchboard.thorben.dk
Have you tried to change the Context on the configurations page?
thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
tgj wrote:
Hi Ronald,
You posted he same question yesterday and I answered you. Do you till have
problems?
Thank you for posting
On more testing, I conclude that Asterisk isn't being very clever about
codec negotiation.
My understanding (possibly faulty) from experiments is this. If I have:
UA1 -- Asterisk -- UA2
and have disallow/allow entries in UA1's stanza in sip.conf, it seems
that the first entry in the allow list
tgj wrote:
Have you tried to change the Context on the configurations page?
yes, ..
please read below
please post an EXAMPLE how you think it works.
thank you
bye
Ronald
thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
tgj wrote:
Hi Ronald,
You
I have a TDM400 with an fxo card installed; zaptel.conf is setup for
distinctive ring, but it only partly works:
Our distinctive rings have the same sound in the very first ring (one long
tone). After that, they differ. As a result, all calls produce a dring
signature of 0,0,0.
If I setup the
Hello,
I had the same problem. I solved it by putting the context of the phone in
sip.phone as the same
context where the hint statement is: i.e.:
sip.conf
[1713]
context=phones
extensions.conf
[phones]
;1713
exten = 1713,hint,sip/1713
exten = 1713,1,Playback(transfer,skip) ; Please
Thorben Jensen wrote:
Version 0.91 - 17. April 2005.
* IPS is now using the context configured in Asterisk for peers - the
context on the configuration page is used for Speed Dial Numbers only
* New tab page for Speed Dial Numbers
Download here: http://ipswitchboard.thorben.dk
Thorben,
I hope
Hi Ronald,
I must admit I am getting confused now.
I understand that you have a problem getting Speed Dial Buttons to work. The
problem as I understand it is that the calls are placed in the wrong
context.
To solve that problem I have asked you to make sure that you have typed a
valid
Thorben,
I hope you find some time to make all more smoothly. It is a great
product, but there are still some unclear things.
Following problems I have encountered:
1. The help system is still in the very first stage, ... a typical
engineer habit ;-)
I hope you can add in the next
On Saturday 16 April 2005 14:00, [EMAIL PROTECTED] wrote:
On Fri, 15 Apr 2005, Paul Hewlett wrote:
I am running * 1.0.6 with 8 analogue phone lines connected to 2 cards -
lspci reveals these as :
The problem is that under certain circumstances (which I am unable to
determine) * bridges 2
Hi ALL;
Itriedto compile res_perl module with
Asterisk, but It failed.I use both Asterisk
and Asterisk-addons lates CVShead.
I did as follows:
1- make a patch to Asterisk Makefile.
2- Try to re-build Asterisk. BUTit
says:
gcc -c perlxsi.c -D_REENTRANT
-D_GNU_SOURCE -fno-strict-aliasing
Is there a working CARD.XML for cisco MGCP phones?
The one on the cisco site is old and it's not working with the new
firmwares.
Thx
Sergio
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On Sat, 2005-04-16 at 13:50 -0700, snacktime wrote:
This also happens to me when I call into my own * box voice system
unless I'm very careful about adding appropriate wait statements after
answering the line. Not sure if this is related to the above problem,
but it made me wonder if an * box
On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said:
Eric Wieling wrote:
I have put into each phone settings (sip.conf and zapata.conf) in my
office:
callgroup=1
pickupgroup=1
I cannot pickup any calls from another phone!!
What do I miss here?
Your SIP phone is
At 05:02 PM 17/04/2005, C F wrote:
Don't use it with queuing, use it with dial
One problem with this: queueing gives a context menu. Just using a series
of 'Dial' commands means that I lose the ability for the caller to have a
context menu without putting a call to WaitExten or Background (both
On April 17, 2005 05:55 am, Tom Fanning wrote:
Illegal instruction (core dumped)
Sounds like you have compiled asterisk for a processor that is greater than
the processor you're running on. I.e. compiled and told it to use P4
instructions when you're on a P3, or maybe even told it to use MMX
Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c and its use
instructions on the wiki, but with no compiling/installation instructions.
Can anybody be of help here?
Thx
Ezabi
signature.asc
Description: OpenPGP digital
FWIW, I finally got it going using queues. The queue has one member which
is a Local/[EMAIL PROTECTED] number. The context it points to uses Dial with the M()
option, and it all seems to work...MoH runs all the time, and the caller
can leave the queue via voicemail.
Thanks for the help etc.
At
With the 2.6 kernel, you can just load ztdummy and not worry about the USB
controller.
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
___
Asterisk-Users mailing list
I still cannot find it:
What does it mean, and how can I fix it?
Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown
SIP command 'PUBLISH' from '192.168.250.108'
I think your phone is trying to do overlap dialing but Asterisk does
not support this yet. I don't think
[EMAIL PROTECTED] wrote on 04/17/2005 11:02:51 AM:
On April 17, 2005 05:55 am, Tom Fanning wrote:
Illegal instruction (core dumped)
Sounds like you have compiled asterisk for a processor that is greater
than
the processor you're running on. I.e. compiled and told it to use P4
I'm in the UK so numbers are generally started with a zero. The
dialstring sent to the sipura is fine, running asterisk
-vvvc gives me called number@101.
Where 101 is the extention number of the sipura.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Is there a Plugin for AMP to ease Polycom 500's Configurations?
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To UNSUBSCRIBE or update options visit:
Hi,
gcc -c perlxsi.c -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -
D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -
I/usr/lib/perl5/5.8.0/i386-linux-thread-multi/CORE -o perlxsi.o
gcc: perlxsi.c: No such file or directory
gcc: no input files
make: *** [perlxsi.o]
Hello:
I read many documents about reducing the codec
bandwidth by 1)compressing the rtp header and
2)implementing point-to-point link. But none of these
documents mentioned how to implement it. So I wonder
why there is not much resources about something
valuable like this which interest many
Hi,
I 'm using a server DL145 with AMD opteron processors, with TE410P
Digium Quad-Span card.
The server is running RHEL4 x86_64.
And have problem to compile codec g729 from
http://www.readytechnology.co.uk/open/g729/,
but ipp sample speech code not problem compile with ia32 or em64t.
use
Send me a copy of your iax.conf and your extensions.conf.
I'll look at it.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of MobilPete
Sent: Saturday, April 16, 2005 6:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] problem
Greg,
Have you checked the 'PSTN Dialing Delay:' setting under the 'PSTN Line'
tab, I suggest this is at least 1 (second), just to let things
stabalise.
% -- SNIP -- %
Greg Wrote:
Lucky you, my spa-3000 likes to dial 911. So far the local cops have
been nice about it though. (my mobile
On Sun, 17 Apr 2005 01:39:09 -0400, Karl J. Vesterling wrote:
H.323 will not traverse NAT.
Sorry... I know, I was a big proponent of it when H.323 was the only
standard VoIP protocol out there. Probably because when it came out NAT
wasn't even thought of.
The problem is that the
This is very interesting to me since I am in the process of setting up SER
to Asterisk in a similar scenario. I'm surprised there haven't been more
posts. Maybe include SER - Asterisk in the title. There are other posters
on the list who use SER and Asterisk together who surely must have
On April 17, 2005 05:55 am, Tom Fanning wrote:
Illegal instruction (core dumped)
Sounds like you have compiled asterisk for a processor that is greater
than the processor you're running on. I.e. compiled and told it to use P4
instructions when you're on a P3, or maybe even told it to
Jesse Guardiani [EMAIL PROTECTED] writes:
Wait a sec... COME TO THINK OF IT!
Why not run asterisk on your linux box that you are running GnomeMeeting
on, and use it to convert between H.323 and IAX and SIP???
After all, it is a penguin...
That's certainly a good alternative. I'm
I keep getting the same answer from people
Well the SIP implementation is fine if you use XXX IP Phone
so obviously Asterisk was never designed to be used as a TDM gateway
but merely as a PBX server only.
On 4/17/05, Cameron Beattie [EMAIL PROTECTED] wrote:
This is very interesting to me
Hello All,
I have been trying a did company for a few days. I find the service
decent, but sound quality only moderate.
Rather than spending 35 or so for monthly with did, I am considering an
isdn bri at this location.
How much more stable and reliable is bri or pri versus a voip did
service?
I have to agree that voipjet is a good service. If only they had did's
it would be even better, but I like the fact that outgoing cid works
well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Greenberg
Sent: Saturday, April 16, 2005 6:35 PM
To:
Hello to all.
I saw on
http://www.intel.com/software/products/cluster/clustertoolkit/features.htm a
software (or feature) a Cluster Toolkit for Linux distributions that use
Intel Pentium 4 Processors.
Does anyone know if is possible to use The \Intel Cluster Software\ for
High
Ed Greenberg wrote:
800 numbers are free to the caller because the recipient pays the charge.
Voipjet has no way to get paid anything for carrying the calls, hence
they are unwilling to use their resources to move calls with no revenue.
Can you blame them? :)
Well it's not a problem, I can
Please share with us the name of the company you had a bad experience with.
Then we can avoid the same problem.
Thanks
Cameron
- Original Message -
From: Linn Boyd [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
No, I suppose you can't blame them
What is FWD's motivation (or IAXtel, etc) to provide this service, then?
-Andy
On 4/16/05, Ed Greenberg [EMAIL PROTECTED] wrote:
800 numbers are free to the caller because the recipient pays the charge.
Voipjet has no way to get paid anything for carrying
Guys.
Ive done some searching and seems that installing spandsp on cvs head had
been a pain because of changes on the patch.
Anybody has a howto on installing spandsp on the recent cvs head? And how
they got receiving and sending faxes worked out?
Hope you can help.
Thx!
Just curious what syntax did you use to load the VMware tools on Fedora
Core 3?
Thanks,
Sean
On Sat, 16 Apr 2005 16:50:56 +0200, [EMAIL PROTECTED] wrote:
I installed asterisk 1.0.7 successfully on VMware workstation with
fedora 3 as guest.
Of course without any hardware only pure asterisk. It
Very true.
I have found the outgoing CID to be very ... useful.
Although occasionally inconsistent on the remote party's end, even
though voipjet's CDR shows the CID string that I sent.
-Andy
On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote:
I have to agree that voipjet is a
On April 17, 2005 05:55 am, Tom Fanning wrote:
Illegal instruction (core dumped)
Sounds like you have compiled asterisk for a processor that is greater
than the processor you're running on. I.e. compiled and told it to use
P4 instructions when you're on a P3, or maybe even told it to
On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote:
Hello All,
I have been trying a did company for a few days. I find the service
decent, but sound quality only moderate.
Rather than spending 35 or so for monthly with did, I am considering an
isdn bri at this location.
Roman Volf wrote:
Have you tried putting both access points on the same channel?
Good suggestion. It now seems to roam between access points nicely, even
while a call is in progress.
Also, I found firmware v1.5.3 if anyone needs it, along with manuals that
are quite a bit more in depth than
Title: a simple question .
I have
successfully installed and configured a WCTE11XP card to connect with a voice
switch (Cisco VCO/4K). Also, I have the SIP connection working as well, where a
call from the voice switch properly transfers to the SIP
phone.
The
voice switch is normally set
Sure sounds like a veiled threat to me. Post something they don't
like and find your support ticket ignored or possibly your account
closed? Oh well guess I won't be getting any support from livevoip
anytime soon:)
Straight from the network status page on their website...
If you are working
snacktime wrote:
Sure sounds like a veiled threat to me.
Veiled? Looks pretty overt to me.
Why do these folks always think they can treat their customers like
, when this is a market that really does have competition? They're
not the incumbents, for God's sake, who get to do whatever they
On April 17, 2005 11:36 am, Philip Warner wrote:
FWIW, I finally got it going using queues. The queue has one member which
is a Local/[EMAIL PROTECTED] number. The context it points to uses Dial with
the M()
option, and it all seems to work...MoH runs all the time, and the caller
can leave
On Thu, 14 Apr 2005 09:33:21 +0500, amna saleem [EMAIL PROTECTED] wrote:
No actually i have successfully installed (from scratch) and been
using asterisk for more than 4 months now...i have been using diax
phone ...but i came across this iaxcomm just thought about
transfering a calljust
Unbelieavable, and utterly disgraceful. Anyone found responsible for
establishing such a policy would quickly find their ass on the street in
any organization that understands the first thing about customer
service. One doesn't build or protect a business by threatening and
bullying one's
I'm curious, how are you licensing your codec? The source is open, but the
codec usage licensing is not. I think you'll find that licensing it from
Digium will be much simpler, not to mention their code will Just Work(tm)
without any messing around.
-d
At 12:08 PM 4/17/2005, you wrote:
Hi,
I
can someone tell me more about this?
On Apr 14, 2005, at 17:55, Roy Sigurd Karlsbakk wrote:
hi
for what I can see on digium's site, there is an x86-64 optimised
g.729a codec. is this particularly optimised for intel or amd? I
wonder most about sse/3dnow/whatever, as AFAICR this is quite
Which file control extension dialing responsivity / timing?
When someone dial my extension, and is not fast enough, asterisk
announces that the extension is not valid (it happened to me too).
I have a mixed of two and three digit extensions in dial plan.
Which setting controls this behavior.
Hi Joseph,
Let me take a guess - the problem only occurs when dialling four digit
extensions?
I think you will find that your dial plan is matching the three digit
extension and then dialling it straight away - Asterisk won't wait for a
timeout before trying to follow the dial plan, as soon as
I think you are right.
I'm using Sipura-3000 which is causing the problem; though I don't know
which setting.
I just double check my dial-plan and I don't have any sort of conflicts:
123 / 1234
where the first three digit would match any four digit in any dial plan.
The problem only occurs when
Hi, I just recently
installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe
application. I don't think it installed with the standard 'make install'
command. If not, how do I accomplish this?
Thanks,
Matt
___
Asterisk-Users
The compilation of codec g723.1 was fine. After I have copied to
/usr/lib/asterisk/modules and started the asterisk -c .. I get this
below error before asterisk quit. Anybody had any idea on Intel codec 723.1
?
[codec_g723.so] = (G723.1/PCM16 (signed linear) Codec Translator, based on
IPP)
Jesse Guardiani wrote:
On Sun, 17 Apr 2005 01:39:09 -0400, Karl J. Vesterling wrote:
H.323 will not traverse NAT.
Sorry... I know, I was a big proponent of it when H.323 was the only
standard VoIP protocol out there. Probably because when it came out NAT
wasn't even thought of.
The problem
Hello all,
I have asked this question of Broadvoice support and the following is their
responce:
John,
Unfortunately we are not able to fully support asterisk. We refer customers
to the Asterisk forums where users are quite well versed and some are
affiliated with BroadVoice.
The only thing
I just want to make the simplest call in which an X-Lite calls another
X-Lite via asterisk. Unfortunately I failed time and time again. If
someone is kind enough to show me sample config files by which asterisk
works well, it will help me a lot.
Best regards,
Abe
On Thu, 14 Apr 2005, Rod Bacon wrote:
I have been frustrated by a variety of zyxel issues/products and have found
the best solution for all of them lies in a cylindrical receptacle that sits
beside my desk...
I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a
built in
On Wed, 13 Apr 2005, Kevin P. Fleming wrote:
[ DELETED]
Realistically, how cheaply can you put together a box with a T-1 card
and a channel bank with 24FXS ports (even disregarding G.729
transcoding, which would add to the cost)? $700? $800? more? I can't say
for sure, but if you wanted
Matt Schwartz wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to
install the MeetMe application. I don't think it installed with the
standard 'make install' command. If not, how do I accomplish this?
MeetMe requires Zaptel. If you do not have Zaptel installed,
There are a couple ways to do this. Or shoiuld be anyway. One is by
setting the context as you have done. The other is by setting the
extension at the end of the register line and doing a
goto(someplace,s,1) for one line and goto(someplaceelse,s,1) for the
other all from the same context.
If
On Sun, 17 Apr 2005, Greg Boehnlein wrote:
On Thu, 14 Apr 2005, Rod Bacon wrote:
I have been frustrated by a variety of zyxel issues/products and have found
the best solution for all of them lies in a cylindrical receptacle that sits
beside my desk...
I've had pretty good luck with the Zoom
So the Panasonic extension dialed by Zap/3/206 command will ring and
Zap/4/221 will not ring at all, even before ext 206 is picked up?
Yes, exactly. Zap/4/221 won't ring at all.
If you have two extensions numbered 211 212, why are you using 206 and 221
in your Dial command?
211 212 is
On Sun, 17 Apr 2005 21:24:30 +0200, Bruno Hertz wrote:
Jesse Guardiani [EMAIL PROTECTED] writes:
Wait a sec... COME TO THINK OF IT!
Why not run asterisk on your linux box that you are running GnomeMeeting
on, and use it to convert between H.323 and IAX and SIP???
After all, it is a
I can call in to and out of * from either number/account that i have.
The problem is i would like to answer with different prompts based on
which account/number the called dialed but broadvoice sends the call as
if it came from whichever account i register second.
Executing
MeetMe is straight forward. Follow the steps for ztdummy and
there you go conferencing
Check out www.voip-info.org for more info
Cheers,
~Vamsi
On 4/18/05, Matt Schwartz [EMAIL PROTECTED] wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to
install the
It would be easier if you could get send us your sip.conf entry and
confiuration made in x-lite
Also, please let us know where exactly the problem is. Is it
while registering the x-lite or during the call and the exact error
messages.
Cheers,
~Vamsi
On 4/18/05, Abraham WEI [EMAIL PROTECTED]
Vamsi Pottangi wrote:
It would be easier if you could get send us your sip.conf entry and
confiuration made in x-lite
Also, please let us know where exactly the problem is. Is it
while registering the x-lite or during the call and the exact error
messages.
Cheers,
~Vamsi
On 4/18/05,
Hi,
Did anyone compare G.729 implementations (from Digium and the one based
on IPP) on features, stability, quality and reliabilty?
It would be intresting to know how they fair against each other.
I could be wrong, but in my testing I did notice a bit more hiss on
Digiums codec
Matt Schwartz wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out
how to install the MeetMe application. I don't think it installed
with the standard 'make install' command. If not, how do I accomplish
this?
Thanks,
Matt
Fixed!
Another way of doing this is to give customer extra seconds between
numbers:
...
exten = s,4,Background(afterhours-menu)
exten = s,5,DigitTimeout,5 ; give them 5 seconds between digits
exten = s,6,ResponseTimeout,10 ; give them 90 seconds to make a choice
...
--
#Joseph
On Mon,
Thank you but... this did not help. the problem is that the calls all come
in
as if from the same account, whichever registers second.
called first number and got:
-- Executing Answer(SIP/xx1492-b2c7, ) in new stack
which should have been xx1405
now that you
Good day all
I installed asterisk on a pc with redhat 9 and a 4port bri
eachtime a call comes in,iax,sip,pstn it just hangs the pc
Top shows 75% of the cpu goes to asterisk?
Any Idea why?
Please Help
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It's safe to assume that this particular company is pretty much
functionally illiterate given the tone and tact of the rest of their
comms. They won't be around long.
On Apr 17, 2005, at 2:58 PM, Rusty Shackleford wrote:
Unbelieavable, and utterly disgraceful. Anyone found responsible for
Hi everyone,
I am trying to figure out a plan for dynamic call forwarding between
multiple asterisk servers. I would be dealing with around 30 different
extension prefixes, each handled by a distinct asterisk server. Is
there a sort of dynamic call routing feature to accomplish this, or I
would
G'day. I've been working with * for some time now, but mostly from a
enterprise perspective. I've just setup my own box at home and want to
enable some more home user type functionality.
Does anyone have a trick to allow the dynamic modification of the
dialplan by users? I want the ability to
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