On the Digium's site it says:
The Wildcard TDM400P is a half-length PCI 2.2-compliant card
while for other cards it says:
The TE411P is for use only with a 3.3 volt PCI slot.
Does the TDM400 not only fits, but also functions in a 3.3V only slot?
From what i detected so far, is that some MOBO
If you do not like giving your SIP credentials to others, you can
install a SIP phone like http://www.pernau.at/kd/voip/ActXPhone/ easily
on your own homepage. It does not allow registration at the SIP proxy,
but this can be added very easily (visual basic).
regards
klaus
kevin ling wrote:
How can I send DTMF from the console?
Anthony.
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Hi,
I have success upgrade two 7960 phone from sccp to sip.
Some tftp server doesn't work. You can try this tftp serverand post your
tftp logs.
http://www.solarwinds.net/Tools/Free%5FTools/TFTP%5FServer/
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
EdwardsSent:
Hi Rich,
Thanks for replying to this question - the decision is confusing me a
lot :)
You said:
"Help us understand exactly what this "incoming traffic flooding the
bandwidth" is suppose to mean. Are you running something else besides web
and voip through this link? If not, then what is
FYI,
http://www.cepstral.com/
You can download the english and spanish voice files for test first. And
modify the festival-script.pl to using cepstral swift program.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Otero
Sent: Thursday, January
On Wed, 2006-02-08 at 12:58 -0500, Matt Roth wrote:
Keep in mind that if you want to run Asterisk Business Edition, RedHat
Enterprise 3 or Fedora Core 3 are currently required in order to receive
full technical support. My options were narrowed down further by the
amount of RAM in our
Hello,
i have set up an asterisk sip to h.323 convertor,
it is working OK. The only problem i have is this :
For example when my identity is [EMAIL PROTECTED] , and i call a sip number
from a sip phone, the called party sees my identity (caller identity) as [EMAIL PROTECTED], which is the
I have defined 4 queue's. Is there any way to check is there any agent logged
in any of those queue's?
What I would like to do is to check if there is any agent in any of queue's and
if there is, then I'll will transfer a call to that queue, it there isn't I
would like to do something else
Hello,
i have set up an asterisk sip to h.323 convertor, it is working OK. The only
problem i have is this :
For example when my identity is [EMAIL PROTECTED] , and i call a sip number
from a sip phone, the called party sees my identity (caller identity) as
[EMAIL PROTECTED], which is the way it
Hi Everyone,
I've got a weird problem with both Firefly iaxLite (both IAX
softphones). They don't seem to stop ringing when an incoming call is
make to them. If the call is answered the conversation starts both ways
but the ringing sound still keeps going and the softphones keep
Hi all,
I have a problem: On my internal S0 where phones are connected via HFC I get
all the number with a leading 0 (either from internal SIP phones or external
dialins via CAPI). I don't know where to look for this 0. Any ideas?
Greetings, Sven
--
Sven Fischer (Dipl.-Phys.) - FACIT
I have a problem with CDR recording in Asterisk 1.2.x. This is the
situation:
An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single
HFC-S ISDN BRI card. I log the call records to both the Master.csv and
MySQL.
The problem is that when an incoming call from the ISDN line is
On Thu, Feb 09, 2006 at 09:12:44AM +, Pete Barnwell wrote:
On Wed, 2006-02-08 at 12:58 -0500, Matt Roth wrote:
Keep in mind that if you want to run Asterisk Business Edition, RedHat
Enterprise 3 or Fedora Core 3 are currently required in order to receive
full technical support. My
Carlos Chavez wrote:
Is there any way to increase the number of digits before the number is
diales automatically?
Yes,
I don't know about the 601s, but under the 301s and the 501s you can
edit the digit map via the web interface or the sip.cfg on your ftp server.
Doug
--
Ben
Dear all,
Does anyone try to install 2 or multiple TE411 card into
one server? Can it be done? What about stability?
Thanks
Ray
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Hi Sven,
Same problem... Not solved...
With CAPI and mISDN.
I think it as to do with
nationalprefix=0
internationalprefix=00
on capi.conf/misdn.conf. I already try to nationalprefix= but always
get that damn 0. If I change nationalprefix=5 I get a leading 5 and so
on... But without any
What have you set the
PSTN Dialing Delay:
on the PSTN Line tab (logged in as admin advanced) ?
Mine is set to 1 and it works well.
Chris
- Original Message -
From: Anthony Rodgers [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
You can even set it to zero. Mine works well when in zero. The line pick up
immediately :
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton
Sent: Thursday, February 09, 2006 6:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I changed these parameters in zapata.conf :
callprogress=no
busydetect=no
And now it's working fine.
Jerome
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jerome
SOUCANY
Envoyé : mardi 7 février 2006 11:04
À : asterisk-users@lists.digium.com
Hi all,
I am trying to setup h.323 connection between two asterisks. The
situation is like that:
asterisk173 only must accept incomming h.323 calls from asterisk172, so
asterisk173 is peer and asterisk172 is user, am I right?
My config files:
Asterisk173:
ooh323.conf:
I had problems with it set to 0 for some reason but that was a very early
firmware for the device.
Chris
- Original Message -
From: Sam Lee [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2006
Hello,
I have a busy problem with Asterisk when I try to transfer a call from PRI
directly to IVR.
This problem appear sometime after 2 hours or 2 minutes.
The log file contain :
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)
When this problem appear I must
Hi all,
I have a strange problem, regarding zap channels and cdr.
I am using asterisk bristuffed version
Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium,
Inc. and others.
with two billion ISDN cards. I also installed asterisk addons, last stable
version via cvs
internal
Can you send some *CLI output? BTW, which spandsp version are you using?
[]'s
MM
Darlon wrote:
I don´t know if the last message was with content. So, I sent again. I have
installed a Digium card TE210P and unicall for use MFC/R2. I think that it´s
all right but I can´t make and receive calls.
Vincent Régnard wrote:
Joseph Rothstein a écrit :
I can't seem to change the default registration for IAX clients:
Feb 6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry:
Restricting
registration for peer 'virbiage' to 60 seconds (requested 3600)
Feb 6 12:23:03 NOTICE[7883]:
Today I noticed that junghanns released the bristuffed version of asterisk
1.2.4 (it was .1.2.2 last week, when I installed)
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1l.tar.gz
Downloading and installing that solved my problem
Andrea
When I try to make att transfer (*2) of call that was in queue the call get's
disconnected. Blind transfer (#1) works fine. In dial plan I don't have any h
or H (hangup call with *). In features.conf I have this line disconnect = *0.
What could be the reason why call hang's up?
--
Tomislav
Hi,
I have defined 4 queue's. Is there any way to check is there any agent logged
in any of those queue's?
What I would like to do is to check if there is any agent in any of queue's and
if there is, then I'll will transfer a call to that queue, it there isn't I
would like to do something
Hi,
Since yesterday my Asterisk 1.2.3 is displaying the
following message every few seconds
Asterisk Event Logger restartedRotated
Logs Per SIGXFSZ (Exceeded file size limit)
This causes my log files (verbose, queue_log) to
become huge with lots of logger rotate messages, but I don't know
Yes, it seems that I was somewhat in error.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
kevin ling wrote:
In my remember, when playback a file. The Asterisk will automatically choose
the audio file with the lowest conversion cost. Not always looks the
filename.gsm.
-Original
Hi,
I've been asked to find a server that is capable of running the Digium
TE410P card. We usually get Dell PE servers and after a quick look I
think the PE 1800 has the required slot:
Six Total: 2 PCI Express (x8 lane x4 lane); 2 x 64-bit/100MHz PCI-X; 1
x 32-bit/33MHz PCI (5v) and 1 x
Hello,
I use this hardware but I have some problems and I don't if these problems
come from the DELL or not.
http://www.digium.com/index.php?menu=compatibility
Regards
Jerome
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Niall Hallett
Envoyé :
I found the problem.
Master.csv reached 2.0GB and since the moment this
happened Asterisk went crazy!
Since I am using cdr-mysql, how do I disable the
use of csvs?
Thank you
Dov
- Original Message -
From:
Dov Bigio
To: asterisk-users@lists.digium.com
Sent:
Hi,
We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where
users register with an OpenSER cluster (2 nodes currently).
When they request PSTN they are forwarded to * where they have entries
in SIP realtime database. This ensures that they get their correct
CallerID and
inline...
You said:
Help us understand exactly what this incoming traffic flooding the
bandwidth is suppose to mean. Are you running something else besides web
and voip through this link? If not, then what is flooding your
bandwidth?
You are right about web page serving not using much
On Thu, Feb 09, 2006 at 10:56:18AM -0200, Dov Bigio wrote:
Hi,
Since yesterday my Asterisk 1.2.3 is displaying the following message every
few seconds
Asterisk Event Logger restarted
Rotated Logs Per SIGXFSZ (Exceeded file size limit)
This causes my log files (verbose, queue_log) to
[guest]
type=friend
context=default
isecure=very
-it doesn;t work , asterisk shows: Feb 9 08:41:13
NOTICE[29683]: chan_iax2.c:6782 socket_read: Rejected connect attempt
from 89.*.8...
for the incoming call
On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote:
Try adding
[guest]
type=friend
context=default
insecure=very
-it doesn;t work , asterisk shows: Feb 9 08:41:13
NOTICE[29683]: chan_iax2.c:6782 socket_read: Rejected connect attempt
from 89.*.8...
for the incoming call
On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote:
Try adding
for sip.conf , there is a configure option for this : allowguest=yes
is there a silimiar setting for IAX ?
On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote:
Try adding insecure=very to the guest user account in iax.conf. This
should not do a user/pass challenge on the incoming call.
Hi Tzafrir,
The problem was the file Master.csv that had reached 2.0GB.
I am writing a cron script to backup this file periodically and prevent this
from happening.
Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Why
don't you log rotate them? or if you do you should do it more
often.
Regards Allan GeePhone: +27 21 4644400 Ext.
103www.equation.co.za
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Dov
BigioSent: 09 February 2006 03:26
Hello,
I might be wrong here, but I thought that in Queues.conf, if you defined a
queue with joinempty=no, or joinempty=strict then no calls will be placed in
the queue, and asterisk will go onto the next extension in the dial plan.
; This setting controls whether callers can join a queue with
Yeah-- sorry...
"
dial-peer voice 635099 voipdescription calls sent
to Asteriskpreference 1destination-pattern
[635-9]..progress_ind setup enable 3session target
ipv4:10.10.1.28dtmf-relay h245-alphanumeric
"
I had been trying to do this with H.323 -- the Call Manager
uses H.323
There are
Hi Henry -
Just started using an asterisk-based PBX with Polycom IP501 phones. Am
Fairly satisfied and am starting to get into FTP setup of the phones.
Have figured out most things except for how button remapping works.
In sip.cfg, I have this entry:
keys
Hi
i would like to start with prepaid and post paid billing system
i would like to have your feed back what all i need to look
and what is the best billing software
where i can configure DID incoming also
some suggestions areapprciated
ram
___
Check your /etc/modules.d/zaptel and make sure you have:
install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg
Sean
Tzafrir Cohen wrote:
On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
Hi i followed this instructions for installing ztdummy on a 2.6 kernel
(taken
Title: Asterisk vs. Traditional PBX
Hi everyone !
So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support
Yeah, I saw that compatibility list and the potential problem with the
onboard ethernet controller. Did you disable it and use a pci based card
instead?
Does anyone else run PowerEdge servers with the TE410P?
Thanks,
Niall
On Thu, 2006-02-09 at 14:24 +0100, Jerome SOUCANY wrote:
Hello,
I
If you
are using the 2620 like a SIP IP-PSTN gateway
your
voip dial-peer would be like this:
dial-peer voice
635099 voipdescription calls sent to Asteriskpreference
1destination-patternT
(or whatever youneed to match)session
targetsip-serverdtmf-relay h245-alphanumeric (or
whatever you
I appreciate the input, but after doing a little research on that card,
it looks like I'll still need the channel bank, I think with some
carefull ebaying, I should be able to do the hardware canceling for
about $1000 less then what i saw the 104d card for.
not to mention it seems total overkill,
Title: Asterisk vs. Traditional PBX
You may be using less then ideal phones. With a Polycom 501, I can't see
you having voice quality issues, With a Sangoma or Digium card and a PRI the
quality and functions of a Asterisk system are on par with most PBX's (I'd say
they're above).
It is a
Hi, Can anyone help me out setting oh323.
extensions.conf
[default]
;Prueba oh323 saliente por timanfaya0h323
exten=_6.,1,Dial(OH323/timanfayaoh323/x${EXTEN},90,tr)
[discriminador]
exten=932289394,1,SetCallerID(932289394)
exten=932289394,2,Dial(OH323/timanfayaoh323/x${EXTEN2},60,tr)
-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 08, 2006 12:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nortel Meridian Opt 81C and PRI
On Feb 8, 2006, at 9:27 AM, Greg Camp wrote:
Hello everyone,
It seems that the letter s did not make it into the original release.
Please visit www.astlinux.org and download the latest tarball.Or,
if you just want s in all of the available formats, just grab this:
http://mirror.astlinux.org/sounds/s.tar.bz2
Sorry!
--
Kristian
Title: Asterisk vs. Traditional PBX
Hi everyone !So here's my question
of the day ! I need to make a decision on whether or not to go to a voip
solution or configure an existing pbx (norstar) that my company has
available. We are a small startup. I'm wanting a solution that will
support up
I've used the spa-1001 and the spa-2001 for faxes. Works good over a
local area network. thevoipconnection sells those for about 60 bucks though.
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000
Tomislav Parčina wrote:
I have running *
Hi,
You just have to remove cdr_csv.so module :
unload cdr_csv.so under the CLI
or add noload = cdr_csv.so in /etc/asterisk/modules.conf and reload
asterisk
--
http://www.olivier-perrin.net
Le jeudi 09 février 2006 à 11:26 -0200, Dov Bigio a écrit :
I found
Tzafrir Cohen wrote:
On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
Hi i followed this instructions for installing ztdummy on a 2.6 kernel
(taken from
http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy
)
cd /usr/src/zaptel
* READ /usr/src/zaptel/README.udev
I've got a telecommuter working out of her home office, using a Snom 200 phone, what happens is occassionally her phone will loose audio one way.She will be talking on a call that was incoming to her extension, and all of a sudden the caller can not hear her any longer, she can here the caller
Hi
All,
I've set up an
Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming
phonenumbers. One phonenumber is for voice-calls, the other one for receiving
faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the
fax-number by G.711. Can I
Niall Hallett wrote:
Yeah, I saw that compatibility list and the potential problem with the
onboard ethernet controller. Did you disable it and use a pci based card
instead?
Does anyone else run PowerEdge servers with the TE410P?
Thanks,
Niall
We run the TE410 on the PowerEdge 1850 and it
Kerry is right on.
We use a similar config in dozens of installs with 200 users and it just
cruises. Consider adding a duplicate server for failover at some point.
-Original Message-
From: Kerry Garrison [EMAIL PROTECTED]
Date: Thu, 9 Feb 2006 06:57:32
To:'Asterisk Users Mailing
we would like to running 4 port or 8 port ISDN BRI card on production asterisk
system.
any one can recommend a good product? is it stable and with good voice
quality?
--
Peng Yong
___
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Well, I don't know what it is at the moment, I just know its a wireless T-1
that I'd migrate over to a different infrastructure.
Actually, TDMoE can route and can go longer distances when you run it over
Mikrotik and use their EoIP. Well, given that the fact that it runs over
Ethernet
Sean Cook wrote:
Check your /etc/modules.d/zaptel and make sure you have:
install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg
Sean
Mine has:
post-install ztdummy /sbin/ztcfg
I changed the line with your sugestion but same result (after reboot):
voip # lsmod
Module
On Thu, Feb 09, 2006 at 09:09:14AM -0500, Sean Cook wrote:
Check your /etc/modules.d/zaptel and make sure you have:
install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg
And what is that good for? You don't need to run ztcfg after loading
ztdummy.
--
Tzafrir Cohen |
On Thu, Feb 09, 2006 at 09:10:26AM -0600, Miguel wrote:
Tzafrir Cohen wrote:
On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
Hi i followed this instructions for installing ztdummy on a 2.6 kernel
(taken from
http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy
On Thu, Feb 09, 2006 at 11:56:02AM -0200, Dov Bigio wrote:
Hi Tzafrir,
The problem was the file Master.csv that had reached 2.0GB.
I am writing a cron script to backup this file periodically and prevent this
from happening.
Any way, if any developers are reading this, I don't think that
Title: Meetme echo cancellation
Hi there
I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not
Invalid module format usually means that the module you're trying to
load was not compiled with the same parameters as the kernel you're
trying to load it into, make sure your /usr/src/linux symlink points to
the kernel you are actually running, /var/log/messages etc. will usually
have more
This is my first foray into SIP telephony, so be gentle. :-)
The Polycom SoundPoint IP 501 phones have been fantastic so far. I still have
a lot to learn when it comes to them, but the manual seems pretty extensive
and so far Asterisk has been playing well with them.
I have a need to be able
Hans Witvliet wrote:
Does the TDM400 not only fits, but also functions in a 3.3V only slot?
From what i detected so far, is that some MOBO manufactures have
pci-slots that provide 3.3 Volt AND 5.0 Volt, thus can handle all kind
of cards.
The TDM400P and TDM2400P will work in any PCI or
Dov Bigio wrote:
Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Maybe a more user-friendly message could be logged, infoming which file
reached the 2.0GB.
Unfortunately when we receive SIGFSZ from the kernel, we
I am thinking of setting up a * system for a remote office in china. I
was going to use a tdm400p to setup a basic 3X8 system.
I will setup the system in the US and ship it over.
Does anyone know of any problems that I should watch out for.
Signaling, caller id, ..
Andres wrote:
We run the TE410 on the PowerEdge 1850 and it is rock solid. We do not
have problems with the onboard ethernet controller.
The interaction between the TE cards and the onboard ethernet controller
affects only a small number of users, and we don't know what
specifically causes it
On Feb 8, 2006, at 1:03 PM, Gerard Saraber wrote:
Hi,
I've had some decent luck with the mark3 echo canceller from the zaptel
driver, echos on about 20% of the calls, people I've called say I sound
great now, but our side hears echos.
I was wondering if there was any way to tweak the current
On Thu, 9 Feb 2006, Peng Yong wrote:
we would like to running 4 port or 8 port ISDN BRI card on production asterisk
system.
any one can recommend a good product? is it stable and with good voice
quality?
I have very good results with Eicon DIVA Server 4BRI
Steven Langley wrote:
I am using IAX2 softphones dialing into a meetme conference. In my softphone
I was forcing uses to click on a button when they wanted to speak, enabling
their microphone and disabling their speakers. This way when a user was
speaking they did not hear their voice half a
Vicidial can't call and transfer to my softphone.
I get some line that says
Spawn Extensionexited on non zero
Here's some of the CLI output. I am using Asterisk 1.2.4 and astguiclient 1.1.8
...thanks for the help
|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and
Andrew Kohlsmith wrote:
Is there a way to get Asterisk to tell the IP501 to use a different ring, put
something up on the display, *something* on a dynamic basis? The wiki
doesn't seem to have a lot of information about this kind of thing.
There are examples (IIRC) of making the phone
Hi all,
getting a server going wiht a few TDM400's and some phones, and some IAXys
too
I haven't heard any issues about AU phones being able to RING in Australia,
like the problem in the UK with ring capacitors on the BT system. Are there
any problems like that?
Also, with the iaxy's -- they
Tzafrir Cohen wrote:
Makefile:204: target `ztdummy.o' given more than once in the same rule.
Something is bad. Did you edit Makefile?
Yes. I delete all the modules, leaving the lines like this
before:
MODULES:=zaptel tor2 torisa wcusb wcfxo wctdm wctdm24xxp \
ztdynamic
With pen in hand, Technical Support succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
There's a vox forum that focuses on Sipuras - post your query there for
good tech help. We've deployed a number of Sipura's and haven't
We have got some ATA for only $55 if you are interested?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Sampson
Sent: Thursday, February 09, 2006 11:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I think this is a
question that has been discussed before.
But you see nowadays most carriers will provide thing like SIP using IP
authorization rather than username and password and I am now wondering whether
Asterisk can do something like that or not?
Sam
On Thursday 09 Feb 2006 16:01, Kevin P. Fleming wrote:
Dov Bigio wrote:
Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Maybe a more user-friendly message could be logged, infoming which file
reached the 2.0GB.
On Thu, 2006-02-09 at 10:05 -0600, Matthew Fredrickson wrote:
Yeah there is, upgrade to trunk and use the new echo canceller there
(MG2). It's supposed to rock, at least from what I've heard. All the
MEC cancellers are _OLD_. At least switch to 1.2 and the KB1 echo
canceler before giving
Good Day,
I have a weird issue using zaptel-1.2.3 and a PRI with 8 voice
channels.
With nobody on the phone using ztmonitor I get the following:
Why would I have such high TX signals on certain channels.
~ron
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 1 -vv
Visual Audio Levels.
Hi Kevin,
I see...
That's why you rotate asterisk logs everytime this message occurs.. it makes
sense.
Unfortunately, in my case, it was the CDR CSV files tha reached that size,
so rotating logs was just worsening my situation, since asterisk started to
generate rotated log files every few
search for:SirrixJunghannsBeronetRegardsRobOn 2/9/06, Peng Yong [EMAIL PROTECTED] wrote:
we would like to running 4 port or 8 port ISDN BRI card on production asterisk
system.any one can recommend a good product? is it stable and with good voicequality?--Peng
hi all a dumb question..
how i do to block the 00 for certain sips extensions?
for example i have the extensions 400 to 500
i need to extension higher than 429 can't digit 00
in my extensions.conf i have
exten = 420,1,Dial(SIP/420,20)
exten = 420,2,Hangup
exten = 421,1,Dial(SIP/421,20)
exten
Yeah, the reported Dell issues seem to be with the x600 series (2650,
1650, etc.) No issues at all on my PE2850s (other than having to talk
Dell into selling me a power cable so the FXS ports would work.)
-Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Can anyone shed some light on what happened?
Asterisk 1.2.1 with Zaptel 1.2.1
Here is what I know happened:
A call came into our main number and was answered
Asterisk set the monitor CALLFILENAME and then started monitor.
The call was directed to a context called open where all calls go
during
Hi Andrew -
I have a need to be able to identify incoming calls based on some factor
(could be time of day, caller ID, dialed number, it doesn't matter.) --
Assuming Asterisk can differentiate between the calls I want, how do I inform
the IP501? There are only three line appearances -- I
The answer is yes, I think, but I don't recall precisely how off the top of my
head, and I'm walking out the door in a moment. The phone will hold more than
a dozen distinct ring tones which you can create for yourself, and you can have
asterisk direct it to use a ring tone independently of
In article [EMAIL PROTECTED],
Steven Langley [EMAIL PROTECTED] wrote:
I am using IAX2 softphones dialing into a meetme conference. In my
softphone I was forcing uses to click on a button when they wanted to
speak, enabling their microphone and disabling their speakers. This way
when a user
Hello every one.I:1. Run install-pdf from linux to support faxes on my asterisk. 2. Made the configurations throuhg AMP in a.Setup-Inbound Routing-(the only route I have)-fax extension-System b. Setup-Inbound Routing-(the only route I have)-fax email-(my email) c. Setup-Inbound
You are in my same situation.
I thought I solved the problem (if you look at tomorrow post) but it isn't
My situation is a bit different: I have the last bristuffed version of
asterisk 1.2.4 (released yesterday)
And I also have 2 zaphfc cards.
but the behaviour is absolutely the same
If you
Hi,
Ronald Voermans wrote:
I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've
configured two incoming phonenumbers. One phonenumber is for
voice-calls, the other one for receiving faxes. I want the incoming
voice-calls to be coded by the G.729 codec, and the fax-number by
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