Yes, i have some but what kind of experience?
Erdem HAKI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ceyhun
KIRMIZITAS
Sent: Saturday, July 23, 2005 12:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk users from Turkey?
On Sun, 24 Jul 2005 17:07:52 +0100, Julian Lyndon-Smith wrote:
I was wanting to stress-test my new server, and as I have a TE410p
card (but only using 2 ports), I was going to connect ports 3 4
with a cross-over cable so that I could make a number of outbound
calls on port 3 and receive them
Difficult to believe that someone worked on any remotely technical
project when he/she relies on AOL for email. Just my 2c. Never thought
anything could be worse than me using Outlook ;-)
-Original Message-
From: Terry H. Gilsenan [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 24,
On Sun, 2005-07-24 at 20:10 -0500, Andrew Latham wrote:
Add your module to your module startup method for your distro. See
http://voip-info.org and search for startup or boot, then read.
Or of course do make config in zaptel, having carefully read
README.udev if running udev.
--
Dave
http://www.junctionnetworks.com
--On Sunday, July 24, 2005 9:20 PM +0200 Marc Storck
[EMAIL PROTECTED] wrote:
Hello,
I'm looking for US DID and US50/CA 800# Providers.
I found voiceconduits.com 8 month ago, there interface looks good, but
there are still not live, I believe they won't be
Many thanks, Niklas.
I'll use this as a basis and let you know how things pan out.
First things first, need to build a E1 cross over cable :)
Julian.
Niklas Larsson wrote:
On Sun, 24 Jul 2005 17:07:52 +0100, Julian Lyndon-Smith wrote:
I was wanting to stress-test my new server, and as I
I dont know if I have the same experiences.
Usually my Skype calls are very garbled at first. I find that my G729 Asterisk
calls are better quality. You can try using ULAW if you have the
bandwidth. It. might make the quality sound better.
Maybe its your SIP client/hardware
phone that
On Mon, 2005-07-25 at 07:43 +0100, Julian Lyndon-Smith wrote:
Many thanks, Niklas.
I'll use this as a basis and let you know how things pan out.
To get a better spread out load (across your various apps) look at the
random application, you should be able to get each call to randomly go
to
does astcc support h323 ,because it doesnt exists h323
in trunks technology.
__
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peiyin wrote:
Dear all,
I want to create a php web front end to disconnect a SIP call (from a
particular sip phone) which is in progress. Any ideas how to do so?
Google for Flash Operator Panel. Or look in the Asterisk wiki for it.
Flynn
___
dbruce ha scritto:
The Cisco 7960 SIP firmware does not support the 7914 sidecar... To
use the sidecar you need to use the SCCP (Call Manager) Firmware.
There is chan_skinny in Asterisk but I don't recall seing support in
it for the 7914. You may want to look at the chan_sccp driver.. it is
Hi everybody,
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain (s${CALLERIDNUM})
exten =
what does your voicemail.conf and sip.conf look like?
Mark
On 7/25/05, Mauro Zanin [EMAIL PROTECTED] wrote:
Hi everybody,
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
Hello,
With the 1.5.2 firmware, have you managed to get one-touch message access when
pressing the Messages button? It worked for me with 1.4.1 but no longer with
1.5.2: I have to go through the message count screen first.
In phone.cfg I have:
msg msg.bypassInstantMessage=1
and in
Hi everyone:
The following code dials our prefix, sends a beep, and sends a DTMF c
tone, then dials the phone number.
I need to send the DTMF only if the phone is answered.
[voip]
exten=i,1,NoCDR()
exten=i,2,Hangup()
exten=s,1,Wait(2)
exten=s,2,Background(beep||)
exten=s,3,DigitTimeout(6)
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
Hi everybody,
Hi Mauro!
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
Title: Outgoing SIP to SER causes LOOP BACK message
Hello fellow asterisk people!
I have Asterisk listening on port 5061 and SER on port 5060.
Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
My problems are with SIP. I can make incoming calls from SIP to asterisk
Kevin Walsh wrote:
Brian West [EMAIL PROTECTED] wrote:
Or better yet.. modify the disclaimer like I and a few others did to
say that the only thing you will disclaim are things you post on the
bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted
under your user on mantis IT IS
Dean Collins wrote:
I've just gotten off a skype conference call and it pisses me off that
the quality of skype is higher than my asterisk calls.
Is there such a thing as a super high bandwidth codec?
Asterisk does not support wideband codecs as far as I know. Most
telephony gear expects
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote:
I don#8217;t know if I have the same experiences. Usually my Skype
calls are very garbled at first. I find that my G729 Asterisk calls
are better quality. You can try using ULAW if you have the bandwidth.
It.
Hi,
we had 3 analog lines but our telco did't pass us the caller-id so
Asterisk tried to identify the caller-id but found bad data. Setting
usecallerid=no avoided the Warning.
Be sure your telco sends you the caller id.
Giorgio
Chris Thompson wrote:
Hey
Thanks for the response but still
Hi,
the option c for the announce of the user count does not work me in * 1.0.9.
exten = ,1,Wait(1)
exten = ,2,MeetMe(|Mdcs)
And how to handel the marked mode with option A? I can't find any sample config
for this.
Regards
___
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain
Hi folks,
In an effort to save bandwidth (our 7905s run over a WAN) we've
switched from ulaw to g729a. We purchased 4 licenses from Digium (4
SIP clients, low call volume), and they seem to have been accepted:
[codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator)
==
Hi,
I get this message after password request in voicemail app:
Unable to create lock file: No such file or directory
Anyone got a clue about fixing that problem ?
I can't understand what directory or file we are talking about ..
Tnx for any help!
--
Best regards,
Alessio
Hi list!
Again Wengo has made changes to their servers that require modifications
to * configs.
Is there anyone that has the 'new' wengo working with asterisk that could
post their configs?
Also they switched codecs, now G720a is required to connect. I can only
find an (open) G729 codec,
We had two cards in the system - BRI card and analog TDM from Digium.
The problem was caused by incorect modules.conf. There was a
post-install directive for the TDM card which runs ztcfg. Additionaly I
was running ztcfg from the asterisk startup scripts as well - which was
the problem.
So I
Caller id is deffinately passed (checked by plugging phone straight into
line).
So its me + asterisk is the problem :) most likely my config.
Cheers
Chris
- Original Message -
From: Giorgio Incantalupo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Kib Eki wrote:
Hi,
the option c for the announce of the user count does not work me in *
1.0.9.
exten = ,1,Wait(1)
exten = ,2,MeetMe(|Mdcs)
That is correct. I was able to get it to work with CVS HEAD.
And how to handel the marked mode with option A? I can't find any
sample
For an Asterisk based "virtual attendant"application thatworks
withlocal inbound calls (in multiple markets) from PSTN phone lines to a
local DIDi.e...Seattle PSTN line call comes into a Seattle DID, or a Phoenix
PSTN line call comes into a Phoenix DIDso all of the activity related
Hi, all
I have problem with echo.
I am running Asterisk server and someone else is running FireFly IAX2 phone.
When I talk to this person I get very strong echo on my end. His end is OK.
At same time, I was trying to set up someone else with exactly same setup
and there is no problem at all.
This may not be the answer you're looking for ... but why
not whip up a little program or web application for the operator's PC that
shows the extension, name, busy status, new voice-mail count, etc? Or
you could have it done by a consultant. In the long run, it will
probably be
Sry for the empty previous reply
Well I think it's not the right way. Using an
external program you will have a delay on lamp status notification and a
overloaded server, manager applications are really cpu power
expensives.
The internal asterisk hint system can immediatly
notify the busy
wassim darwish wrote:
does astcc support h323 ,because it doesnt exists h323
in trunks technology.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hello!
Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl
over POTN.
Yesterday i configured asterisk to be able to hear a song (with Playback()
command) with
iaxcomm, and it was wonderful.
Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a
Also they switched codecs, now G720a is required to connect. I can only
find an (open) G729 codec, is this the same as G729a?
I only have it working one-way, no incoming calls. Ironically, when
Mark was here we caould have gone to meet them and straighten it out
once and for all :)
Dear users,
I am new to this mailing list. Can someone send me a guide
or steps to configure Asterisk on Linux box? I will highly appreciate.
Regards,
Afzaal
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Yes and ilbc is more robust against packet loss, jitter etc. with
using not very much but less more bandwith. Asterisk has support for
ilbc and there are many providers offering PSTN termination with ilbc
codec. And voice quality is better than g723. check out
http://www.ilbcfreeware.org/
On Monday 25 July 2005 14:07, Afzaal Mirza wrote:
I am new to this mailing list. Can someone send me a guide or steps to
configure Asterisk on Linux box? I will highly appreciate.
http://www.voip-info.org
___
Asterisk-Users mailing list
Hi Afzall,
i am also still a beginner on *. A made best experience with the * wiki on
http://www.voip-info.org/wiki-Asterisk. Maybe you start with the introduction part.
Afzaal Mirza wrote:
Dear users,
I am new to this mailing list. Can someone send me a guide or steps to
configure
As promised, here's one I wrote myself...
http://www.systemhalted.com/zish/bharni.mp3
It should make for very appropriate hold music for Public Broadcasting,
PBS and the like.
On Sun, 2005-07-24 at 17:44 -0400, Steve Gladden wrote:
OK was actually able to pull it out of the archives!
It's
I believe that this can help you on your question.
http://www.automated.it/guidetoasterisk.htm#_Toc49248757
Regards,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Monday, July 25, 2005 8:22 AM
To: Asterisk Users Mailing List -
update your CVS - recompile and try again
cheers,
Mark
On 7/25/05, Alessio Focardi [EMAIL PROTECTED] wrote:
Hi,
I get this message after password request in voicemail app:
Unable to create lock file: No such file or directory
Anyone got a clue about fixing that problem ?
I can't
Deniz Pecel wrote:
Yes and ilbc is more robust against packet loss, jitter etc. with
using not very much but less more bandwith. Asterisk has support for
ilbc and there are many providers offering PSTN termination with ilbc
codec. And voice quality is better than g723. check out
Hi,
Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per
http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone
The dialplan was configured through AMP and has nothing fancy in it.
As a first time user of not only Asterisk, but also a PBX, there are
some operator teething
Steve Kennedy wrote:
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote:
I don#8217;t know if I have the same experiences. Usually my Skype
calls are very garbled at first. I find that my G729 Asterisk calls
are better quality. You can try using ULAW if you have
Rudolf Ladyzhenskii wrote:
Hi, all
I have problem with echo.
I am running Asterisk server and someone else is running FireFly IAX2
phone. When I talk to this person I get very strong echo on my end. His
end is OK.
At same time, I was trying to set up someone else with exactly same
setup
Hello,
I have configured my polycom ip600 and ip500.
The phone works well.
But the clock is wrong and flashes the whole time. Drives me nuts!
I have set the time offset on the DHCP / boot server. 36000 (I'm in
Australia!)
It doesn't seem to help...
Mike
Steve Underwood wrote:
Steve Kennedy wrote:
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote:
I don#8217;t know if I have the same experiences. Usually my Skype
calls are very garbled at first. I find that my G729 Asterisk calls
are better quality. You can try
I have a Dell SC400 on the shelf, and I'm considering just installing
Asterisk and two TDM04B cards and shipping it down. Does anyone have
multiple TDM cards in the SC400? FXO ports on a TDM card are about
$75/FXO port.
I have not run your Dell hardware, so I can't comment on that.
Please could someone help?
I need a way of knowing the DNID for a call that is in our queues.
I've tried setting varies variables with different _ __ before them but
none are inherited by the Local/ call that is placed from Queue.
Any suggestions appreciated.
Neil
Thanks for reply.
:)
1. Ask if they are using a speaker and mic built into the PC (This will
create echo) - solution: Tell them to use a headset
2. Check if they have any output volume (in volume control, advanced,
recording) set to record.
3. Check if they have a crappy sound card -
Nathan Pralle wrote:
However, for FXO ports, I'm using the Digium Wildcard X100P's which can
be obtained on eBay for $9-$20, usually. Much cheaper price-per-port,
although the TDM would give better expandibility.
You mean NON Digium X100P's. Digium no longer sells the X100P. The
cheap
As nice as HDLC sounds in theory; we have the same setup, a T1 with afew
lines split off, and i just don't see a need to add the routing load to
the asterisk machine. We have an Adtran 604 which splits the T into a
PRI and 10/100. Incidentally, HDLC in asterisk seems to be.. a hassle
to get
With the 1.5.2 firmware, have you managed to get one-touch message
access when
pressing the Messages button? It worked for me with 1.4.1 but no
longer with
1.5.2: I have to go through the message count screen first.
In phone.cfg I have:
msg msg.bypassInstantMessage=1
In the phone.cfg
Hi,
I would like to use a digum card to call an external number through my
PSTN. I think that I have a problem in the configuration. Asterisk
returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap'
I use Fedora core 3.
I installed libpri, zaptel and asterisk.
I plugged my
ACK! I forgot about that. You're right, my bad. Still, they seem to
work well.
Nathan
Eric Wieling aka ManxPower wrote:
Nathan Pralle wrote:
However, for FXO ports, I'm using the Digium Wildcard X100P's which
can be obtained on eBay for $9-$20, usually. Much cheaper
price-per-port,
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote:
With the 1.5.2 firmware, have you managed to get one-touch message access
when
pressing the Messages button? It worked for me with 1.4.1 but no longer
with
1.5.2: I have to go through the message count screen first.
In
Eric Wieling aka ManxPower wrote:
Steve Underwood wrote:
Steve Kennedy wrote:
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote:
I don#8217;t know if I have the same experiences. Usually my Skype
calls are very garbled at first. I find that my G729 Asterisk calls
Hi Mike -
I have configured my polycom ip600 and ip500.
The phone works well.
But the clock is wrong and flashes the whole time. Drives me nuts!
I have set the time offset on the DHCP / boot server. 36000 (I'm in
Australia!)
It doesn't seem to help...
Sounds like you're running the 1.5.2
Hi all.
When I place a call Im getting this error:
Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on
'UniCall/13-1', but no exception handler
Lots of this messages appeared on my Asterisk full log and the CPU
got 100%.
Topology:
Analog Phone - Hicom300H E1 - E1 Asterisk -
There should be a NTP setting.
Setup Network Time Protocol.
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Title: Soft Phone
Firefly Third Party version beats all others.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: Friday, July 22, 2005 4:12 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Soft Phone
Can anyone
Yes, I have that setup too (no change from 1.4.1)
Are you saying one-touch voicemail works for you with 1.5.2 ?
(meaning no message count summary screen when pressing Messages)
If you have more than one line, you will get a summary, otherwise, how
would you access the other lines
Hello all,
I wondering if you guys could provide me with some advise. We have
developed a simple management tool for a small site which is
architected in the following way:
Server A == Server B == Server C
Server A is an asterisk server with a TE410P which connects to the
PSTN as well
On Sunday 24 July 2005 22:44, Steve Gladden wrote:
OK was actually able to pull it out of the archives!
It's now at http://stuff.michiganbroadband.com/asterisk
I'll leave it there for about a week or two then remove it.
T othe best of my knowledge it's public domain, if anyone needs more
Steve Underwood wrote:
Eric Wieling aka ManxPower wrote:
Do yu have a link for wideband-ilbc info?
It is described on the GIPS site, along with the narrow band ilbc. The
wideband one is not offered to the world on a royalty free basis, as the
narrow band one is. I have never looked at how
On Mon, Jul 25, 2005 at 02:07:54PM +0200, Paolo wrote:
oh: I use i4l asterisk driver because i need to use isdnlog, and i
dont know if other drivers allow me to still use it .
During my tests i tried to kill isdnlog (i dont know if it may block the
devices) but i got always those errors
On Mon, Jul 25, 2005 at 03:14:44PM +0300, Deniz Pecel wrote:
Yes and ilbc is more robust against packet loss, jitter etc. with
using not very much but less more bandwith. Asterisk has support for
ilbc and there are many providers offering PSTN termination with ilbc
codec. And voice quality is
Lucky you - i have had 2 die before i opted for the TDM.
- Original Message -
From: Nathan Pralle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 2:40 PM
Subject: Re: [Asterisk-Users] Need
Title: SER Asterisk SIP =513 Message Too Big
Using Asterisk 1.0.9
When I try to make an outgoing call with SIP I get the message 513 Message too big back from SER. Any ideas what I am doing wrong?
Debug below.
SER and Asterisk are running on the same Server
SER is on port 5060
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
Brian West [EMAIL PROTECTED] wrote:
Or better yet.. modify the disclaimer like I and a few others did to
say that the only thing you will disclaim are things you post on the
bug tracker! NO UPDATES, NO CHANGES, NO
Well you were right, and I don't understand why that exact config worked for
my iax config, but doesn't work for my sip config. Maybe they don't permit
as many variables in the sip file, who knows.
I ended up using:
[frombroadvoice]
exten = _X.,1,Noop(Incoming call for extension ${EXTEN}in
On Mon, Jul 25, 2005 at 03:44:07PM +0200, Alexis F. wrote:
Hi,
I would like to use a digum card to call an external number through my
PSTN. I think that I have a problem in the configuration. Asterisk
returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap'
I use
Louis-David Mitterrand wrote:
On Mon, Jul 25, 2005 at 09:38:29AM -0400, Noah Miller wrote:
With the 1.5.2 firmware, have you managed to get one-touch message access
when
pressing the "Messages" button? It worked for me with 1.4.1 but no longer
with
1.5.2: I have to go
Dear All,
Sorry to be posting again. I have solved my problem.
The problem is that when exiting from the macro, the priority number
is still in effect. For example, priority 1 is at the start before
entering macro after the macro the priorty will be 2. Since there
isn't any other dialplan
Dear friends,
please excuse me if my question will be trivial.
I've installed and started Asterisk (stable 1.0.7, but with CVS HEAD
I experienced just the same problem), and changed a bit sip.conf:
[general]
; ...
dtmfmode = inband
disallow = all
allow = ulaw
allow = alaw
allow =
Get [EMAIL PROTECTED] here:
http://asteriskathome.sourceforge.net/
This should be the EASIEST first time install out there. Once you get
familiar/comfortable, consider building your own following steps at
http://www.automated.it/guidetoasterisk.htm
-Original Message-
From: Afzaal
Adam Dobrin wrote:
As nice as HDLC sounds in theory; we have the same setup, a T1 with
afew lines split off, and i just don't see a need to add the routing
load to the asterisk machine. We have an Adtran 604 which splits the
T into a PRI and 10/100. Incidentally, HDLC in asterisk seems to
For AAH or AMP support you might have better luck searching the forums at
these locations:
http://sourceforge.net/projects/asteriskathome/
http://sourceforge.net/projects/amportal/
And I'm assuming you have already been to:
http://asteriskathome.sourceforge.net/
you can find a summary of codes
I'd say your hardware is out of codecs. Sipura SPA-2000's, for example,
only allow one G729-call at a time because of licensing issues. Allow
GSM as a secondary codec and you should be fine. Yep, it's more
bandwidth, but...
-Original Message-
From: Andrew Furey [mailto:[EMAIL
Hello All,
I have new TDM04B installed and working fine with Asterisk 1.0.5 built on
RedHat 9.
All is working fine except CallerID that bothers me big time.
I have several Panasonic and Sony phones and CallerID works fine with it
(when I plug in the line into phone instead into Asterisk I
Hi
Was wondering if anyone could confirm this?
agent is logged in using agent callback login to a sip extension
when placing a call with the API action: originate and the channel: agent/blah
the call only works
one time then seems to tie up the sip channel.
when placing a call using a call
[EMAIL PROTECTED] wrote:
There should be a NTP setting.
Setup Network Time Protocol.
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Afzaal Mirza wrote:
Dear users,
I am new to this mailing list. Can someone send me a guide or steps to
configure Asterisk on Linux box? I will highly appreciate.
Regards,
Afzaal
I would like to use a digum card to call an external number through my
PSTN. I think that I have a problem in the configuration. Asterisk
returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap'
I use Fedora core 3.
I installed libpri, zaptel and asterisk.
I
I have asterisk FXO module on a TDM400P
hooked to an Intertel Single Line Card. I can place Intertel intercom
calls to Asterisk (both SIP and analog phones) and the reverse, but transfering
calls doesn't work. Here what the transfer looks like:
Intertel FXS Transfer Asterisk
FXO Asterisk FXS
Hi Alexis,
Quoting Alexis F. [EMAIL PROTECTED]:
I would like to use a digum card to call an external number through my
PSTN. I think that I have a problem in the configuration. Asterisk
returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap'
I make modprobe zaptel
Clearly that's not the answer I'm looking for. I need a hardware phone
for this person. As the following post points out, a software solution
is less than ideal for many reasons. Not the least of which is my user
does not have a PC (please God, don't suggest that I buy a PC).
I need a phone with
Hello
I am using a Junghans quadBRI ISDN card and it is
loaded and working. In Asterisk if I connect to ISDN line it is detected
and tells me so.
In my zapata.conf I have
(abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel = 1-2
;plus group
http://www.globalipsound.com
Try there.
/b
On Jul 25, 2005, at 8:15 AM, Eric Wieling aka ManxPower wrote:
Steve Underwood wrote:
Steve Kennedy wrote:
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen
wrote:
I don#8217;t know if I have the same experiences. Usually my
On Mon, Jul 25, 2005 at 06:38:30PM +0400, Timur V. Elzhov wrote:
Dear friends,
please excuse me if my question will be trivial.
I've installed and started Asterisk (stable 1.0.7, but with CVS HEAD
I experienced just the same problem), and changed a bit sip.conf:
[general]
; ...
One way would do for me, I only use wengo for my outbound calls since they
are a lot cheaper than our Royal Dutch KPN :)
Which codec did you use and could you post your config lines?
Thanks!!
Remco
On Mon, 25 Jul 2005, Wilson Pickett wrote:
Also they switched codecs, now G720a is required
Any recommendation for Linux environments (without WINE) ?
Thanks
Alex
On Mon, 2005-07-25 at 10:04 -0400, Kanuri, Seshu (Company IT) wrote:
Firefly Third Party version beats all others.
__
From: [EMAIL PROTECTED]
Hi all,
Does anyone know a way of playing a sound while the dial command is
running? I want to play a sound every 10 seconds while relevant phone(s)
are ringing and have ring tones played to the caller in the gaps.
I am aware of the fact that you can play a class of MoH during the dial
but I can
U can set your linux to do this work. An SNTP Server.
Bruno.
Billy Dunn wrote:
[EMAIL PROTECTED] wrote:
There should be a NTP setting.
Setup Network Time Protocol.
___
Hi
On Mon, Jul 25, 2005 at 10:31:07PM +0800, Kenige Ho wrote:
Dear All,
Sorry to be posting again. I have solved my problem.
What is there to be sorry about? Posting a summary of the solution is
quite useful, and is definetly considered sound rather than noise.
Now when someone searches the
Hi all,
I am fairly new to asterisk, i have been having this problem for
the past few weeks. When i press *62 (wake up call), it was working
fine last time, but now, it goes like, Please enter the time for
your wake up.. and stops there.
How do i debug this problem?
Thanks,
Hello !
I would like to get working a Fritz PCI card using chan_misdn
operating in ptp mode.
Afer compiling mISDN into the kernel and building chan_misdn
Asterisk stops loading with :
[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri))
== Parsing '/etc/asterisk/misdn.conf': Found
I have new TDM04B installed and working fine with Asterisk 1.0.5 built on
RedHat 9.
All is working fine except CallerID that bothers me big time.
I have several Panasonic and Sony phones and CallerID works fine with it
(when I plug in the
line into phone instead into
Asterisk I get
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