Hey everyone,
Hopefully someone can point me in the right direction for this.
Currently we have two offices, all using Polycom 601 Revsion E I think.
All have the same configurations and firmware versions.
The differences:
Office A: public IP address.
Office B: NAT (router has a static
Friends in the Asterisk community,
Yesterday the Asterisk development branch, also known as svn trunk,
changed quite a lot. We added
two major features: A new version of AEL and a new configuration
system. Hang on, and I'll explain!
* AEL - The Asterisk Extension Language
On Tuesday 25 April 2006 00:57, Juan Salas wrote:
Hello.
Has anybody knows how run two asterisk process
in one hardware? (each one with its own configuration?)
It is possible.
1) Use different UDP ports for SIP/IAX/RTP
2) Use different log files and astdb files
But most users do not need
Scheda wrote:
I'm sure this has been asked a million times. Therefore, I must ask
again. Generally speaking, what do you guys think of it. It looks pretty
good, but for my uses, I'm not sure that a calling card method is the
*best* way to go. But, either way, what is the general concensus?
On Tue, 25 Apr 2006, Dmitry Ivanov wrote:
On Tuesday 25 April 2006 00:57, Juan Salas wrote:
Hello.
Has anybody knows how run two asterisk process
in one hardware? (each one with its own configuration?)
It is possible.
1) Use different UDP ports for SIP/IAX/RTP
This is not necessary
Greetings to all,
I ma having a problem with channel variables on a couple of our Asterisk
boxes.
Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our
external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN.
On the External GW, we also have an IAX trunk to a VOIP
Hi all
I think this could be en old question. I would like to do a
realtime billing prepaid system, mainly using asterisk.
I have found few things;
I can not get CDR function into agi because asterisk set them
once the call is absolutely finish (at least main values
Hi,
I only check the AAH AMP. The inbound routing from-pstn didn't include
the context ext-group. So the ring group setting doesn't work when you
call from PSTN.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo
Sent: Tuesday, April 25,
Yes, just set the hotline number to an extension number. And disable the
welltech IVR function.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: Tuesday, April 25, 2006 3:51 AM
To: Asterisk Users Mailing List - Non-Commercial
25 apr 2006 kl. 00.24 skrev Thomas Winter:
Am Monday 24 April 2006 18:39 schrieb Doug Lytle:
Thomas Winter wrote:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the
Display
Name) for the initial INVITE to an SIP proxy.
If I use SET(CALLERID(name)=) the display-name is
HiI sent this earlier, but it was late and I haven't saw any reply. Maybe now I will have more luckDoes anyone know the correct settings of zapata.conf
and zaptel.conf that are needed to connect two asterisk boxes over E1. I am trying to (just for testing purposes) connect two * ( A and B ) boxes
Hello,I get an Error every minute on the second card of two installed TE410P Cards in our System.The error is: PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8)PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8)Is it possible that there are known problems with 2
I have been writing a lot of AGI programs in C with good success.
I would like somehow to have an AGI program continue in the background
while the pbx execution returns to the dialplan and continues. Is this
possible? I was thinking that perhaps I could fork or create another
thread within the AGI
hmm.. does really nobody had such an issue before?
Thomas Artner wrote:
Hi!
I am using asterisk with two tdm400p cards.
Sometimes (one call out of ten), when a call comes in and is taken,
there is some terrible noise for a short time in the line (for about a
second).
Both partys can
Am Tuesday 25 April 2006 11:24 schrieb Olle E Johansson:
25 apr 2006 kl. 00.24 skrev Thomas Winter:
Am Monday 24 April 2006 18:39 schrieb Doug Lytle:
Thomas Winter wrote:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the
Display
Name) for the initial INVITE to an SIP
Hi Tony
I have the same problem you have, i think what would you like
to do (as me), is to update in a realtime basis credit for prepaid
customer, look what I posted today,
its from ramcluster and the threat is billing realtime, this is what
i discover right now.
Hope it help you
Hi:
I upgraded SuSE to 10 and Asterisk to trunk and now
after deleting all modules and previously compiled
stuff and recompiling asterisk, zaptel, and libpri, I
get this failure of asterisk to start:
[pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]:
loader.c:726 __load_resource: new style
I've been trying
lots of configurations now. And the problem that I can't solve is
this:
I have a Digium
T205P card. I have connected one of the connections to our internal PBX (NEC
2000 IPS). The Asterisk is configured as pri_cpe, and the NEC is configured to
be the network side of the
Hi
I am trying to use festivall with asterisk , I am
using RHEL4 , asterisk1.2.7.1 and festival-1.95-beta
, I am able to hear the voice form the text file ,
when I dial to the extension, but when I have , in
my text file , it plays only the text upto ,
and in the CLI , the , is
Hi
We have some users who are supplying
very long, broken contact details (from Cisco 7912 phones):
Apr 25 11:29:46 WARNING[1480]
chan_sip.c: No closing bracket found in '1st Floor Scanner - 137 sip:[EMAIL PROTECTED]:5060;user=phone;transport='
Apr 25 11:29:46 NOTICE[1480]
chan_sip.c: '1st
Mike,
As someone else mentioned, the delay in getting the disconnect from the
CO is a function of the CO equipment and there isn't much you can do
about that. In one of my test cases from yesterday, disconnect came
within three seconds of the pstn phone hanging up.
I'd have to guess that
Hi Olle,
Very well, but can we do for you during the french day in Paris and what are
the conditions ?
I have announced your event near to all of our resellers/fitters in our
country.
I have talked about that event until our African contacts ;-)
Best Regards,
Francois BERGERET.
Mike Fedyk wrote:
Rich Adamson wrote:
Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding
from scratch. I installed FreePBX (CentOs) from scratch and asterisk
was running, but had not yet been configured. It too crashed with a
kernel panic. Ran memtest for 24 hours; no
Title: Running commands from dialplans
Hi,
Can anybody tell me which version of
res_perl I have to install on Asterisk 1.2.4.
I tried to compile res_perl version 3.5 on
Asterisk 1.2.4 and I got the following error.
gcc -Wall
-DRES_PERL_BASE=\/usr/local/res_perl\ -DMULTIPLICITY -
Hi,
I have 2 installs complaining of the same problem. They are both using
Asterisk 1.0.10. They complain that when someone leaves a message, they
are being cut-off. We tried playing with the maxsilence,
silencethreshold and maxmessage without sucess.
Any hints?
Thanks,
Andre
I think my DTMF problems are solved, but the solution isn't crystal
clear. I reverted back to 1.2.6 and then had the idea to have asterisk
email me every time someone hit the invalid extension. The email
contained the number they dialed and the channel (read sipura box) they
came in on.
Olle E Johansson wrote:
25 apr 2006 kl. 00.24 skrev Thomas Winter:
Am Monday 24 April 2006 18:39 schrieb Doug Lytle:
Thomas Winter wrote:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the Display
Name) for the initial INVITE to an SIP proxy.
If I use
Hi,
What is the version number of the lastest stable release, and how to get
it through CVS or wget? Thnx.
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25 apr 2006 kl. 15.34 skrev Wai Wu:
Hi,
What is the version number of the lastest stable release, and how
to get
it through CVS or wget? Thnx.
All of the information you look for is easily available on
http://www.asterisk.org
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk
Ok... I am not a telephone guy... I was born after rotary phones, so
forgive my ignorance in this matter. I am trying to get a really old
rotary phone up and running with an ATA. Why? Who knows... just
thought it would be cool. The problem is that it does not have an RJ11
connector, instead
Yellow=ground - not used
Green = tip
Red = ring
connect green/red to rj pins 4/5
You could pick up a quarter mod line cord (mod to spade) and replace
the cord, or use a screw terminal block to connect to line.
Enjoy
On Apr 25, 2006, at 9:19 AM, Sean Cook wrote:
Ok... I am not a
Red and Green are the Tip and Ring. Yellow may have to be strapped to
one or the other depending on the phone you have.
Some phones may not ring at all, due to special frequency ringers
installed in them for party lines.
Western Electric did not use these.
As to the ATA, MOST ATA's do not
Hello
I'm using voicemail with realitime. And I need use two diferent
and separate databases.
thanks.
jsalas
-Mensaje original-
De: Mike Fedyk [mailto:[EMAIL PROTECTED]
Enviado el: Monday, April 24, 2006 8:24 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto:
hugolivude wrote:
Funny you mention that Kevin. I was on the web site this morning and
I saw it here:
http://www.digium.com/en/products/hardware/analogcards.php
Later on the same day, that page had changed. The text was gone and
the TDM2400P TDM400P had swapped positions...
I'll
Hello,
Has anyone Knowledge about softphone IAX for pocket PC totally free?
Tkanks for all.
--
Sandra Salmerón Ntutumu[EMAIL PROTECTED]
Tlf. Analog: +34 914888405 / Móvil: 653574298
Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010
Fundación EHAS: Enlace Hispanoamericano de
Jerry Jones wrote:
Yellow=ground - not used
Green = tip
Red = ring
connect green/red to rj pins 4/5
You could pick up a quarter mod line cord (mod to spade) and replace
the cord, or use a screw terminal block to connect to line.
Enjoy
This worked perfectly! Thank you!
Sean
On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote:
This worked perfectly! Thank you!
Sean
Now, I think the question is, does your ATA actually support
rotary/pulse dialing? Mine (SPA-2000) did not. I bought a (very cheap)
MITEL-1 Smart Dialer and went through a RIDICULOUS amount of pain
trying to
Unless you have a top of the line Pocket PC don't even bother. Most
inexpensive units like the T-Mobile MDA just dont have the processing power
to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I
forgot about already and the sound quality was horrible regardless of using
I do have a TDM400 and the Sangoma A200. I have done pulse with the
TDM400, but have not with the A200. I have just never seen a phone like
this... ;)
Rusty Dekema wrote:
On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote:
This worked perfectly! Thank you!
Sean
Now, I think the
After lots of testing I discovered that I could get the sound to work. The only
thing I had been testing was MeetMe and Voicemail. But when I dialed a
SIP-phone, or routed back to other phones via the PRI interface, everything
works just great! The problem only seem to occur when dialing
Rusty Dekema wrote:
On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote:
Now, I think the question is, does your ATA actually support
rotary/pulse dialing? Mine (SPA-2000) did not.
Most/all of the SIP based ones seem not to.
I bought a (very cheap)
MITEL-1 Smart Dialer and went through a
Hi,
If I am interfacing with a legacy PBX system a few questions.
#1 What do I need to do to configure 1 port on a dual port card as
pri_cpe and another as pri_net? Do I just change my config half-way
through the zaptel.conf file?
#2 When I setup span=1,1,0,esf,b8zs doesn't the esf
Sean Cook wrote:
I do have a TDM400 and the Sangoma A200. I have done pulse with the
TDM400, but have not with the A200.
The A200 works with pulse dial.
If yours does not, contact Sangoma or use the latest drivers. They fixed
it after I contacted them several weeks ago.
John Novack
On 4/25/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Unless you have a top of the line Pocket PC don't even bother. Most
inexpensive units like the T-Mobile MDA just don't have the processing power
to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I
forgot about already
Add an Answer() as your first step in your dialplan and see if that
help.
snip
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Same results here with my PPC-6700 nice phone no processing power, I
found that my EVDO card on Laptop works great with SIP softphones.
Unfortunately, I have to agree. I was very pumped about being able to
use VoIP over WiFi on the PPC-6700 (which has a 416 MHz cpu), but the
phone's
I use IaxComm with good results on axim x51
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kerry Garrison
Sent: Tuesday, April 25, 2006 11:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] About
Also note that the Smart Jack allows the Telco to provide T1
Signalling in places that it couldn't in the past, most smart jacks
that I have used are:
[CO]-Optical-[Hut DMS]--[Hut Smart Jack]-HDSL-[CPE Smart Jack]
For the list, Telco Techs, mostly do as they are told, and are
schooled by the
Robert Augustyn wrote:
I use IaxComm with good results on axim x51
Is that something you developed yourself? If so, can you share it? For
the last year I have been trying to find time to get iaxcomm working on
a WinCE machine.
Regards,
Steve
Hi,
I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5
and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash
button. A hook flash works fine for setting up a 3way call. But pressing
the flash button doesn't do anything. The zapata config is below. Anyone
Awhile back I remember someone posted a SMS to DISA AGI script. I searched
the archives and found nothing...anyone out there remember?
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Can't you do all of this with the (Absolute) time setting? So if the
person has 4,000 minutes left.. set the call length for 4,000 minutes
as the absolute max. Alternately... you could probably use screen?
Launch an AGI from the main AGI using screen so it goes into the
background...
You
Hi Steve and everyone
I have a very strange problem with an old 1st gen TDM410P card I've been
using in the production machine (10-15K calls/day) without problem with
Asterisk 1.0.7+Unicall 0.0.2.
When I switched to Asterisk 1.2 and Unicall 0.0.3 (even in 1.2.7.1 and
pre9), span 3 handled mfc
Does anybody have a working Asterisk server with Unicall using MFCR2
in Brazil? Were having problems. It seems SPANDSP never detect the
tones from the telco. Im using brazil protocol variant. Im having
lots of problems
to find out why spandsp seems to not detect the MF tones. We send the
first
Steve,
I a sorry, I should have verified what I am writing.
The software is PPCIAX2 and you can find it: http://www.voipalia.com/ppciax/
Is it pretty not but it works.
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Underwood
Sent:
I've already tried that, but the result is the same... :(
I've also seen the same error reported a long time ago, on this link:
http://lists.digium.com/pipermail/asterisk-users/2004-August/053365.html.
But I can't find a solution anywhere...
Best regards,
Peter Olsson
Visionutveckling AB
i have a client that
wants a function that will automatically logout an agent from a queue if they do
not answer a call. This would prevent future calls from being sent to that phone
if the agent forgot to logout. Any ideas?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros -
Well it works! The pulse detection is a little squirrelly, even with
the debounce changes to wctdm.c. I can't get an audible ring but it
does work.
Sean
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Use the local channel to call the agent first, and if there is no answer, log
them out.
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Tue 4/25/2006 2:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
Hi All I want tosetting as belows.caller --- call ( from telco) -- asterisk --- IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between caller and SIP 2 (don't used call conference) SIP3 want to hear
Hello,
I patched asterisk patched with the latest t38 support
.
I would need some people for tests.
Regards
harry
___
Faites de Yahoo! votre page d'accueil sur le web pour retrouver
Issam,
Don't mean to press, but did you have a solution or a similar experience?
- Original Message -
Date: Tue, 25 Apr 2006 01:37:18 +0100
From: issam [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Channel Restart and Dropped calls
To: Asterisk Users Mailing List - Non-Commercial
Quick question:
Is there a way to distinguish between calling MSN's when using chan_misdn?
More info:
I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base
number plus 5 MSN's. Now I want to my * to do different things when
receiving a call on from different MSN's (like
Sean Cook wrote:
Well it works! The pulse detection is a little squirrelly, even with
the debounce changes to wctdm.c. I can't get an audible ring but it
does work.
Sean
By audible ring do you mean you can't get the phone to ring?
If that is the case, and you have tried connecting the
Does anyone know how to make fastagi continue to the next priority if it can
not connect to the remote AGI Server? Right now I am just getting Hangup and
cant find anything on the net about this.
Thanks,
Steve
___
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Hello,
I hope someone who has been successful in getting Plus.Net's VOIP service
to interface with Asterisk might be able to help.
For some reason I can't seam to register or make outgoing calls. If
anyone would mind posting their register line as well as the Plus.Net
context in the
On 4/24/06, Chris Gamble [EMAIL PROTECTED] wrote:
We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are
getting frequent restarts on the spans which lead to dropped calls. I have
pasted some hopefully
maybe this is related:
On a TDM2400 with 3
FXO modules, is there a way to split each line into basically being its own
trunk or another way to pull off the following scenerio:
PBX has 12 inbound
PSTN lines
1,3,5,7 are the 714
phone number hunt group
2,4,6,8 are the 888
phone number hunt group
9-12 are fax
We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user
Yes, that is the functionality I am looking for, just not sure how exactly
to pull that off.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Tuesday, April 25, 2006 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Hi Geoff,
You might want to try tcdump, specifying the source
and destination IP (to minimize the info)
and see where are the RTP packets going ;
youwill see if they change port or
something like that
after a while.
Cheers,
Frederic
- Original Message -
From:
Geoff
If I remove the eco cancellation module from a TE411P card, will it
work as a plain TE410P?
--
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
signature.asc
Description: This is a digitally signed message part
Carlos Chavez wrote:
If I remove the eco cancellation module from a TE411P card, will it
work as a plain TE410P?
Yes. You can also the 'vpmsupport=0' module parameter to disable the use
of the module without physically removing it.
___
Is there a way to send a caller directly to the queue member if one is
available, without answering it and putting him on MOH? Note that
playing a fake ringing tone instead of music does not suffice, I
really want to not answer it.
The problem: it´s very hard to convince a client that a caller
Via dialplan maybe?
exten = xxx,1,Dial(SIP/101_Queue,20,tr)
exten =xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1)
Kerry Garrison escribió:
Yes, that is the functionality I am looking for, just not sure how exactly
to pull that off.
_
From: [EMAIL PROTECTED]
Ok, anyone used Sphinx with Asterisk? The docs are great at telling me how the
internals of the damn thing work, but now how to USE it. I can't find a single
example of how to run 'decode' in command line mode, without specifying a
billion options!
Doug.
that is a nice function
I use a cronjob to logout everyone each evening if anyone wants that script I would love to provide it.
On 4/25/06, Alberto Sagredo [EMAIL PROTECTED] wrote:
Via dialplan maybe?exten = xxx,1,Dial(SIP/101_Queue,20,tr)exten
Im experiencing touch tone recognition issues when
calling some outside phone systems. For instance, if I call my Nextel phone, and
try to press * to enter my voicemail, Nextels system does not hear
the DTMF tone. Ive also experienced other outside phone systems for
which I am unable to
These people usualy hang out in the downtown area every morning
waiting for work. Do you plan on actualy frying them and then transmit
them over your fax machines using FoIP? You should check with your
local government if it's legal. If you use just their photos then you
might need written
I am asterisk newbie, so please bear with me if this
is an easy one. I am trying to enhance a basic calling
card application to support the feature where the
caller can press ## to end the current call and return
to the main menu to place a new call. Any hints as to
how to go about doing that? I
Anybody have suggestions on having a 56K dialpool and VOIP
connections with an Asterisk box over the same set of PRIs? We've
done the PM3 with PRIs for just dialup, but are looking for a way to
integrate our Asterisk box and move our voice calls onto the same PRIs.
Ian
--
Ian White
I experienced this today. Doing a 'show channels' in Asterisk showed a
Zap line perpetually ringing the sip phone even though the sip phone was
reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip
allowed 2-way audio to resume.
Phil
Frederic Jean wrote:
Hi Geoff,
You
Ian White wrote:
Anybody have suggestions on having a 56K dialpool and VOIP connections
with an Asterisk box over the same set of PRIs? We've done the PM3
with PRIs for just dialup, but are looking for a way to integrate our
Asterisk box and move our voice calls onto the same PRIs.
No
I've been asking about this problem in Asterisk channel... I didn't report it has a bug...Probably it is recommended... On 4/24/06, Thomas Winter
[EMAIL PROTECTED] wrote:Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta:
How do I report a Bug to Digium? or asterisk project?Did you report this
A Lucent MAX TNT will do it, there are some limitations on the TNTs
ability to received caller ID name from the telco if is not sent as part
of the ISDN SETUP message, many Telco's send CNAM in the FACILITY IE and
the lucent ignores it.
-Original Message-
From: [EMAIL PROTECTED]
Ian White wrote:
Anybody have suggestions on having a 56K dialpool and VOIP connections
with an Asterisk box over the same set of PRIs? We've done the PM3 with
PRIs for just dialup, but are looking for a way to integrate our
Asterisk box and move our voice calls onto the same PRIs.
There are
Is there an easy fix?- Dan L.On 4/25/06, Philip Edelbrock [EMAIL PROTECTED] wrote:
I experienced this today.Doing a 'show channels' in Asterisk showed aZap line perpetually ringing the sip phone even though the sip phone wasreset a few times.Doing a 'soft hangup' on the stuck Zap and the Sip
Steve, you need the FastAGI contingency patch, part of the Asterisk
Queues Tutorial available at
http://www.orderlyq.com/asteriskqueues.html
It's near the bottom of the page.
Anybody know why this still hasn't made it into trunk?
Matt.
How can I do the following 2 things in my dialplan?
1. find out what extension a agent is assigned to by agent id.
2. find out what agent is assigned to a extension by extension id.
Anybody know how to do this? I read some where that I might have to pull it
from the db. Example code is a
Bryan Mahin wrote:
I’m experiencing touch tone recognition issues when calling some
outside phone systems. For instance, if I call my Nextel phone, and
try to press * to enter my voicemail, Nextel’s system does not “hear”
the DTMF tone. I’ve also experienced other outside phone systems for
Has anyone actually used these USB speakerphones
http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
Seems to get a pretty good review here
Here I have attched file of extensions.conf please if any one have the soluition to face the problem of voice brakage. My internet E1 data line connectivity is okbecause when i wont use asterisk server then voice is clearThank YouShobhit Nirala+919871476403 SHOBHIT NIRALA CONT NO. 9871476403
It's all possible.
Paul Hales
--
Paul Hales
Technical Manager
Asterisk IT
bus: 03 8320 8100
mob: 0434 225 491
Crazy Boy wrote:
Hi Friends,
I want to implement VOIP PBX service in my office. I have 10 computers
and a server. All computers are Pentium IV processors with 512 MB RAM.
All
I am running AAH 2.8. I have an IVR for our main phone number that
allows users to dial an extension directly. I would like to have a
this call may be recorded announcement played before the call gets
transferred. There is not a built-in option for this in the IVR web
interface, but one way I
John Novack wrote:
Mike Garey wrote:
well, the problem isn't that the card doesn't detect a disconnect,
it's that it doesn't detect it immediately (or at least within a short
period).
Odds are that is the telco, and not the Sangoma or Digium card. That is
quite normal for a 10-30 second
Are you in the USA or Canada?
Mike Garey wrote:
yes, I'm using kewlstart
On 4/24/06, Sean Cook [EMAIL PROTECTED] wrote:
On Mon, 2006-04-24 at 17:20 -0400, Mike Garey wrote:
As far as I can tell, after discussing this matter with other asterisk
users in my area, my telco _does_ provide
Hi,
due to the fact that different providers need a different way to dail,
I made some mistakes, whenever I changed the code.
Multiple gateways, different dialing patterns,
I would need some testing numbers in different countries. Testing
numbers where a tape is or where a long
Which version of unicall and spandsp are you using? How is your
zaptel.conf and unicall.conf?
[]'s
MM
-Original Message-
From: Moises Silva [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Tue, 25 Apr
How about using time announments? I list of these
for each country would be great!
Jason
Ronald Wiplinger wrote:
Hi,
--snip--
I would need some testing numbers in different countries. Testing
numbers where a tape is or where a long company announcement is.
Do you know such numbers?
On 4/24/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi Friends,
[..snip..]
--- Employee 1 PC (Softphone i.e., Headphones with Mic)
--- Employee 2 PC (Softphone i.e., Headphones with Mic)
--- Employee 3 PC (Softphone i.e., Headphones with
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