Yeh problem is they are directly buying from providers in US/UK without
paying 12 % tax on voip .. i guess people who buy itsp license can resell
this minutes by paying tax to government in between .
On 08/12/06, ram [EMAIL PROTECTED] wrote:
I'm not sure, but does this only apply to VoIP
Vicky wrote:
I have around 20-30 softphones behind NAT .. My sip.conf has
nat=yes and
they all are able to register and make calls with no problem . My
voip
carrier supports gsm as well as ilbc .. Server takes calls from sip
phones ,
does call recording in between and forwards to
canreinvite = yes in sip,conf ( trunk section ) ??
No t,t in dial command . No call recording in between , same codec should be
supported by both trunk as well as extension . If trunk is iax2 and
extension is sip then also asterisk will sit in media path .
On 08/12/06, Alex Guan [EMAIL
Yeh asterisk seems to use extension number for calls between extensions on
same server and sends callerid only for outside numbers ( via sip trunks ) .
On 08/12/06, Greg Kennedy [EMAIL PROTECTED] wrote:
I have a site running asterisk 1.2.8 with a hand full of polycoms and
grandstream 2Kxp's.
Dovid B wrote:
tacking pn = adding on - sorry for not being more specific.
I have seen that people in the past have used a linksys router to run
asterisk. It would be to expensive to bring in a PC for every
location. So we want to import cheap home routers put asterisk on
them as use them as
Hi all,
I'm looking at some suggestions from you techies out there.
Let me explain my scenario. Im a reseller to callshops.
I need to take around 100 concurrent calls. Almost all endpoints are sending
G723 codec and my peers take G729.
Can anyone recommend the Server Specs that is ideal for
Hi All
Can anyone suggest a comprehensive GUI manager for Asterisk. It doesn't
matter if it is open source or commercial.
We currently have 100's of users currently managed via the real time
database. Groups of users belong to their own contexts.
We would like a system that is able to
Doug Crompton wrote:
John,
Two questions on your comments
I have no seen an Insteon computer controller similiar to the old bottle
rocket. Is there such a device? I am thinking of getting an Insteon
starter kit bit I have so many X10 devices it will be awhie before, if
ever, that I get
Hi Scott...
http://www.bicomsystems.com/products/
Senad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Pinhorne
Sent: 08 December 2006 11:00
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Management GUI
Hi All
Scott,
What you write sounds standard to any Commerical Application.
Our Call Center version has much more besides:
CallCenter:
http://87.238.74.83/admin/
[EMAIL PROTECTED]
pbxware
I will contact you directly if I might.
Steve
steve 'at} bicomsystems .dot} com
- Original Message -
As I understand your configuration , dial-peer voice 697617664 voip, only
forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX
.115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your
Asterisk box.
An incoming call in your E1 must much a destination
Hi all,
Thanks for your reply,
I'm using sip communicator(in java that is intergrated with one ERP ) and
asterisk is interfaced with this.
i'm able to make calls between pingtel and Voip user,
and also i can able to make call from Sip communicator to pingtel or Voip
phone.
but now i'm can't make
Update on this -
I tried with the newest spandsp on the snapshots site still to no
avail. I also ensured no other copies of spandsp exist, and adding
SPANDSP_LIBS=-lspandsp to makeopts, but still getting the segfault
when rxfax is called.
On 07/12/06, Matt Gibson [EMAIL PROTECTED] wrote:
Doug,
The Uniden CLX465 supports stutter dial tone (SDT) and provides a MWI.
Might be overkill since it is an answering machine as well. There are a
few others. Google for stutter dial tone or phone company compatible
voice mail. The SPA3K can produce SDT. The Budgetone 102 also has an
MWI.
The only time I have seen this problem myself is when Asterisk (and
therefore rxfax) was built when the wrong spandsp header/library files
were present on the system.
The required order of events is:
1) Build spandsp
2) Install both spandsp binary libraries and includes, ensuring no old
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Another solution is to use the Pickup() command. It will pick up a call on a
specific extension that is in the ringing state:
[Description]
Pickup([EMAIL PROTECTED]): This application can pickup any ringing
channel
that is calling
http://pastebin.ca/271763
Hi to all,
To Fran:
As I understand your configuration , dial-peer voice 697617664 voip, only
forward the pattern 697617664( destination-pattern 697617664) to
XXX.XXX.XXX. 115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I
think is your Asterisk box.
you
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4
To make it more understandable - Cent OS 4.4 doesn't have problems with Zaptel
installation.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello,
We have several regional asterisk's connected to a central one making
the the PRI calls through a TE410P card.
When using SetCallingPres(prohibited) on a call at the regional level,
that setting it not forwarded to the
[EMAIL PROTECTED] a écrit :
Hi all,
I'm looking at some suggestions from you techies out there.
Let me explain my scenario. Im a reseller to callshops.
I need to take around 100 concurrent calls. Almost all endpoints are
sending G723 codec and my peers take G729.
Since Digium doesn't
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone
of
Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs
great on it. Debian is good too. They have Asterisk packages, but they're
yum can be used...
direct download from
http://isoredirect.centos.org/centos/4/os/i386/CentOS/RPMS/
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of
Red Hat Enterprise Linux, so
Hi
i am planning to develop a php script that will be called from AGI for
the management of an IVR application.
I'd like to be able to do the following things from php:
- retrive callerid
- play some audio files to the caller
- wait for some DTMF digits
- retrive the DTMF
- stop the call
the
Yes, as long as Asterisk is in between the two, it can perform the
protocol translation.
regards
David
On 12/8/06, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Users..
Is it possible to do. one UA is SIP and other UA is IAX2,
UA(sip)---OpenSER-- Asterisk-- UA(IAX2)
.
Use an empty line key to monitor the other phone
On Dec 7, 2006, at 1:44 PM, Bill Gibbs wrote:
Figures I email this and realized I can hit
Menu
1 (Features)
4 (Presence)
2 (Buddy Status)
Wow that’s a lot of key strokes. Anyway to reduce that to a one
button touch? I don’t mind
This should get you started:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
http://phpagi.sourceforge.net/
Regards,
Ove
nik600 wrote:
Hi
i am planning to develop a php script that will be called from AGI for
the management of an IVR application.
I'd like to be able to do the following
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA)
I just signed up to test their service and they sent me a Number, Proxy, port
and password.
Every reference I have tried leaves me with a 404 error coming from Vonage.
If you have a working setup, please post some
Gracias por el translate!
2006/12/8, Angelito Manansala [EMAIL PROTECTED]:
IN ENGLISH VERSION:
Good night I have mounted the system of predictive marker ASTGUICLIENT in
2 Servants, in one of them who are a Server HP Proliant G3 350 3GB ram with
11 Slackware and
g723 codec isn't problem, you can obtain for all asterisk versions from:
http://kvin.lv/pub/Linux/Asterisk/
PJ
Jean-Michel Hiver wrote:
[EMAIL PROTECTED] a écrit :
Hi all,
I'm looking at some suggestions from you techies out there.
Let me explain my scenario. Im a reseller to callshops.
how can protocol translation affect jitter propagation to both voip ends
(UAs) for dejjiterring? because iax doesn't use RTP for voice stream, it
can be issue (?)
PJ
David Thomas wrote:
Yes, as long as Asterisk is in between the two, it can perform the
protocol translation.
regards
David
John Novack wrote:
Carla Schroder wrote:
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't
run
into some gotcha down the road where there is some
David Cook (Canada) wrote:
On 12/7/06, Dovid B [EMAIL PROTECTED] wrote:
Hi list,
Can anyone who has successfully ran asterisk on a home router please
give
me the modell number as well as how they did it ?
Thanks.
Dovid
Sure. I have 5 units out there on Linksys WRT54GS v1.1
If you are new to CentOS or redhat based OS's, I would recommend using
yum, as it will resolve any dependencies automatically.
If you wish to install RPMS directly, you can download them from any
CentOS mirror. See the CentOS website.
Note: a default install of CentOS installs a bunch of
Hi all,
I have a problem with dialing digits from my analog phone connected to
TDM400 with one FXS card. I can call the phone from SIP, but when I try
to dial digits from it, after first digit I receive a busy tone. I
thouht that it is the problem with DTFM frequencies, so I changed zone
to
Jon, I would be as well very interested in your Voicemail Solution :
AGI + Web Interface to retrieve voice messages.
By the way, you sotre to MySQL, do you use ODBC for that ?
or something else, in that case, what ;o) ?
Thanks in advance !
Jean-Marc
On 12/7/06, Jon Farmer [EMAIL PROTECTED]
Nothing is end to end in this case.
It is two separate sessions, one SIP and one iax.
--
--
Steven
http://www.glimasoutheast.org
Pavel Jezek [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
how can protocol translation affect jitter propagation to both voip ends
(UAs) for
nik600 wrote:
Hi
i am planning to develop a php script that will be called from AGI for
the management of an IVR application.
I'd like to be able to do the following things from php:
- retrive callerid
- play some audio files to the caller
- wait for some DTMF digits
- retrive the DTMF
- stop
/macro-text |
mailx -a /var/spool/asterisk/monitor/20061208-103611:1001.gsm -s
hello [EMAIL PROTECTED]) in new stack
-- Executing Hangup(SIP/1001-081d9b80, ) in new stack
== Spawn extension (rec-tt-trunkdial, *912126245943, 7) exited
non-zero on 'SIP/1001-081d9b80'
Nothing actually
Hi list,
I need no control a call via AMI or AGI or whatever. I don't know how to put
a call on hold.
Example: an external call ring, in the dial plan I call Dial application
to an internal SIP phone. But my SIP phone does not have the on hold
feature, so how to put the callee on hold ?
One suggestion is to transfer the call to an on-hold extension that
plays music, then go pick up the call later... or get a new SIP phone.
: )
~Joel
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Duchatelet
Sent: Friday, December
so that, jitterbuffer should be enabled forced on sip and iax channel
on asterisk (because UAs have no knowledge about jitter on opposite
link), from first example?
UA(sip)---OpenSER-- Asterisk-- UA(IAX2)
Steven wrote:
Nothing is end to end in this case.
It is two
currently running on pbx (pid = 1999)
Verbosity is at least 3
-- Hungup 'IAX2/voicepulse02-8'
-- Executing Wait(SIP/1001-081d9b80, 2) in new stack
-- Executing System(SIP/1001-081d9b80, cat /etc/macro-text |
mailx -a /var/spool/asterisk/monitor/20061208-103611:1001.gsm -s
Does anyone have personal experience with a 5.8gig wireless phone (system)
that has an MWI that WORKS with asterisk via fxs (in my case spa3k)
generated MWI. I know the spa3k does stuttered dialtone but not sure if it
generates FSK MWI.
I see some that state they do but I also see reviews that
Doug Crompton wrote:
Does anyone have personal experience with a 5.8gig wireless phone (system)
that has an MWI that WORKS with asterisk via fxs (in my case spa3k)
generated MWI. I know the spa3k does stuttered dialtone but not sure if it
generates FSK MWI.
I see some that state they do but I
BerkHolz, Steven wrote:
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA)
I just signed up to test their service and they sent me a Number, Proxy, port
and password.
Every reference I have tried leaves me with a 404 error coming from Vonage.
If you have a working
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VOIP PBX) 1-866-638-1254
(Voip PBX) Free World DialUp: 780-217
WebSite:
Another way would be to control the channel from asterisk.
It is a SIP feature, not an asterisk feature.
I have a SIP phone (not a softphone) and want to control it from the
computer.
Greg
One suggestion is to transfer the call to an on-hold extension that plays
music, then go pick
The service is Business Plus. It is a BYOD SIP service.
--
--
Steven
http://www.glimasoutheast.org
Paul [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
BerkHolz, Steven wrote:
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA)
I just signed up to
Hi all
I have installed asterisk 1.2.13
on my P4 Pc with 512MB Ram , FC5
Trunk with my sip provider, on the provider side
i have purchaged g729 installed
on the client X-lite using speex
when i try to make call, i in the log below message
Dec 8 12:18:52 WARNING[5216] codec_speex.c: Out
Hi Jerry, THANKS A LOT.
I viewed configuration files so many times, but I had to be blind so I
didn't noticed that mistake. I was solving this problem for almost two days
with no success... thanks a lot again. :)
It can sound weird, but I cannot wait for Monday when I go to work... :D
Petosh
Has anyone been able to get Asterisk to work with Verizon's VoiceWing
service? I'm in the process of testing Asterisk to see if it will fit
the needs of my company. Since I already have Verizon's VoiceWing
VoIP service, I figured if I can tie into it, that would let me
evaluate service
Hello Users..
Is it possible to do. one UA is SIP and other UA is IAX2,
UA(sip)---OpenSER-- Asterisk-- UA(IAX2) .
UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ).
other wise we can like that..
UA(SIP ) --- Asterisk-UA(IAX2)
But SIP message and IAX
That and any other ref.s I have found give me a 404 error when dialing out.
My Sip show registry is also empty.
ref:
We're at 64.x.x.x port 12146
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726) to
In this case, the machine was a spandsp virgin, it had never been
installed before.
I made sure I ran ldconfig before and after building, and still no joy.
I have managed to get iaxmodem and hylafax to work quite well though :-)
Chris
On Fri, 2006-12-08 at 12:43 +, Steve Davies wrote:
The
I understand this function (line 832 in
app_voicemail.c) is used to retrieve a voice message.
What I don't understand however is why .txt is
appended to the end of the filename. Could someone
shed some light on this for me?
Thanks,
Jez
if (msgnum -1)
make_file(fn, sizeof(fn), dir, msgnum);
Other than for Zap cards, why would you want to switch from *BSD to linux?
I don't run * on *BSD, but I've heard it runs very smoothly and stable
(probably more than several linux distros).
Just curious.
Thanks,
Daniel
-Original Message-
From: John Novack [EMAIL PROTECTED]
Sent: Thu,
This may be a Linux newby question, but here it goes.
I was reading the instructions on downloading and installing Asterisk GUI, but
I can't get this to work.
svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
What would be the equivalent command in CentOS 4?
One thing you could do is use a third-party product like our QueueMetrics
(available free for smaller systems/SOHOs) and use its own internal logic
to link a callerid to all other information (call status, agent, time,
etc), search by different criteria and remote call listening.
Hope
Hi all,
I'm using Asterisk 1.4.0-beta2 and lately I've noticed that I'm having trouble
accessing my voicemail at work using phones on my Asterisk system.
I have to press the * key during the voicemail login process. When I do, it
seems that Asterisk eats it and doesn't send it along.
I
On 8 Dec 2006, at 15:02, Jean-Marc Salsa wrote:
Jon, I would be as well very interested in your Voicemail Solution :
AGI + Web Interface to retrieve voice messages.
By the way, you sotre to MySQL, do you use ODBC for that ?
or something else, in that case, what ;o) ?
Thanks in advance !
svn is application called subversion, you should download and install it
first.
- Original Message -
From: Ed Nuñez
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, December 08, 2006 7:18 PM
Subject: [asterisk-users] downloading asterisk GUI
This
I have 4 Polycom phones with multiple line keys so multiple incoming
calls work fine
The way I would like the incoming call flow to work is as follows:
1) 2 groups consisting of 2 phones each
2) Incoming call rings the first group, if no answer, the 2nd
group is rung
3)
I try to use the background cmd for send incomings call on dial
plan.
I try in an internal number for resting:
exten = 405,1,DigitTimeout,5
exten = 405,2,ResponseTimeout,10
exten = 405,3,Background(vm-accueilcreat)
exten = 1,1,Goto(creat-in,s,1)
exten = 2,1,Dial(IAX2/301,15,tr)
exten =
On Fri, Dec 08, 2006 at 08:23:36AM -0800, je . wrote:
I understand this function (line 832 in
app_voicemail.c) is used to retrieve a voice message.
What I don't understand however is why .txt is
appended to the end of the filename. Could someone
shed some light on this for me?
This is
that site also has g729 codecs for asterisk but is it legal to use them ?? (
digium charges $10 each g729 channel )
On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:
g723 codec isn't problem, you can obtain for all asterisk versions from:
http://kvin.lv/pub/Linux/Asterisk/
PJ
Jean-Michel
On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote:
Hi Steve.
Thanks, but unfortunately, I can't be involved in that. We are
running Asterisk in a production environment and we're using
1.2, not 1.4. I don't have the resources to work with 1.4.
that site also has g729 codecs for asterisk but is it legal to use them ?? (
digium charges $10 each g729 channel )
On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:
g723 codec isn't problem, you can obtain for all asterisk versions from:
http://kvin.lv/pub/Linux/Asterisk/
PJ
Jean-Michel
Hi,
Have anyone experience repeated digits when connecting a call from SIP and
terminating it to a PRI Channel? On the other side of the PRI Channel is an
IVR that expect a pin but the digits come repeated. For example, you dial
12345 but it is received as 12224445
--
Gustavo Flores
IT
I'm having trouble with Polycom 501 phones that asterisk forgets how
to reach them.
/etc/asterisk/sip.conf:
[general]
context=default
MusicOnHold=default
port=5060
bindaddr=0.0.0.0
srvlookup=no;yes
language=en
dtmfmode=rfc2833
maxexpiry=600
defaultexpiry=120
[502]
type=friend
username=502
How long in seconds is the vm-accueilcreat recording?
Have you tried pressing 1,2, or 3 while it's played?
On 12/7/06, Olivier Saulnier [EMAIL PROTECTED] wrote:
Hello,
I try to use the background cmd for send incomings call on dial plan.
I try in an internal number for resting:
exten =
I'm having trouble with Polycom 501 phones that asterisk forgets how
to reach them.
...
host=dynamic
We've found much better results with the static IP here.
Can you try this?
--
Henry J. Cobb
http://www.io.com/~hcobb/
___
--Bandwidth and
-Original Message-
From: Steve Murphy [mailto:[EMAIL PROTECTED]
Sent: Friday, December 08, 2006 12:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Douglas Garstang [EMAIL PROTECTED]
On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote:
Hi Steve.
Steve Murphy wrote:
*snipped
I've been fixing manager bugs here and there, and am willing to take on
any manager issues out there, for 1.4, and trunk, especially, so as to
have things nice and solid for 1.4 before it gets out of beta.
*snipped
Richard-- I'll lab up
1.4 and see if I can get
yum install subversion
On 09/12/06, Kovar Petr [EMAIL PROTECTED] wrote:
svn is application called subversion, you should download and install
it first.
- Original Message -
*From:* Ed Nuñez [EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial
callerid=John Doe 1234
On 05/12/06, Sven Beisiegel [EMAIL PROTECTED] wrote:
Hi...
I just started working with Asterisk and found something that looks
like an error, but i want to be sure, so that's why I'm asking you.
When i make a call from A to B (both SIP clients), I don't see the
name of
Great, exactly what I was looking for. Thanks so much!
Shabbat shalom
Jez
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Dec 08, 2006 at 08:23:36AM -0800, je .
wrote:
I understand this function (line 832 in
app_voicemail.c) is used to retrieve a voice
message.
What I don't
Hi,
Which is the best book to self-learn SIP ?
Regards
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,
Has anyone managed to compile app_nvfaxdetect on asterisk 1.4?
Is there any other way of detecting incoming fax calls on non-Zap channels?
Julian.
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asterisk-users mailing list
To
Derek Whitten wrote:
John Novack wrote:
snip
That sounds like a microsoft way of doing things.. install 25X more crap than
you will ever use. What ever happened to planning and RTFM?
I guess it all depends on what the objective is.
One can sit around and RTFM and play with oneself
David Thomas wrote:
If you are new to CentOS or redhat based OS's, I would recommend using
yum, as it will resolve any dependencies automatically.
If you wish to install RPMS directly, you can download them from any
CentOS mirror. See the CentOS website.
Note: a default install of CentOS
Does there seem to be a popular Linux distro folks use specifically for
Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with
Linux distros. In particular, I'm looking for a free, stable, well
supported distro that has a friendly community. Any advice appreciated.
CentOS
I´m looking for the same feature performed with the manager, but I think
should be the same problem you are experiencing
I need to place music on hold (park) an specific call, while the agent
performs a process/question/inquiry, and then retakes the call.
Is there not a way to park the call?
Thanks
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail list
Sent: Friday, December 08, 2006 4:08 PM
To: Asterisk Users
So there are 0 watchers while the GXP is configured to that hint? are you
sure you set the phone to Asterisk BLF?
On 11/15/06, Ken Williams [EMAIL PROTECTED] wrote:
Upon further investigation I must be doing something wrong.
It was my understanding that a hint extension could be anything, it
Hi
Ok I have the right version many thanks
However I am still a tad stuck (Sorry)
I have all the configs to upgrade from SCCP to SIP
but what config files do I need just to upgrade the sccp to the 7.0-3 version.
I am assuming I need to have a file in the tftp dir that tells the phone to
load
Anyone else have problems with soft buttons not being responsive at all? 2
of the 4 soft buttons do not respond, no matter how hard you push. It is an
IP500. Well over 1 year old.
___
--Bandwidth and Colocation provided by Easynews.com --
the autolifter is for phones without a headset jack.
On 12/7/06, J. Oquendo [EMAIL PROTECTED] wrote:
Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360),
I noticed my client is having some form of feed back on the phone.
Because of Snom's inner oddities this is how I got
On redhat based OS's I would do this...
You can run the following command to see what services are enabled:
chkconfig --list | grep 3:on
Then disable whichever ones you dont need... The services may vary a
bit depending on hardware or what packages you have installed.
I often disable
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Hash: SHA256
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Version: 9.5.1 (Build 1557)
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Hi
Does trixbox comes with a predictive dialer, i want to use a predictive dialer
with trix box or asterisk, please let me know what is the best tot use.
Regards
Kanishka
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asterisk-users
Normally when you think of using Bluetooth with mobile phones you think
of using it to attach a headset wirelessly to a mobile phone... can it
work the other way? Can I have a Bluetooth card on my laptop/desktop
such that my mobile phone can be a handset to a softphone on the
laptop/desktop?
Hi all,
We are in the process of setting up a E1 (TE110p)connection based asterisk
server in which we want to record all the voice conversations.Is this facility
supported on asterisk if so how to configure.What are hardware dependencies
invloued in setting up this facility.
Asterisk can record all outgoing calls ( see voip-info.org for asterisk cmd
monitor and mixmonitor ) hardware requirements depends on volume of calls to
be recorded . Faster sata raid or scsi drives recommended for high number of
alternate calls .
On 09/12/06, Raja Chidambaram [EMAIL PROTECTED]
Thanks, but unfortunately that is an expensive 2 line phone compared to
others in their line that have a base and two or three remotes for the
same price. Seems a lot to pay for a MWI.
I wonder if anyone has had experience with panasonic wireless 5.8gig and
MWI?? They advertise compatibility on
You're trying to teach a pig to sing. The uniden items you refer to
probably have their own internal answering machine, mine does. It's
designed to light the lamp only when it's own machine has a message.
On 12/8/06, Doug Crompton [EMAIL PROTECTED] wrote:
Thanks, but unfortunately that is an
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