Hi,
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 4:59 AM
Subject: [Asterisk-Users] IAX clients and the flash button
Hi guys
As usual I am playing around with IAX soft clients. I was wondering with
the various IAX clients, IAX
Hi
I installed a x100P card today. Once it is configured * no longer starting.
It gives me the following error.
== Parsing '/etc/asterisk/zapata.conf': Found
WARNING[1074432736]: File chan_zap.c, Line 629 (zt_open): Unable to specify
chan
nel 1: No such device or address
ERROR[1074432736]:
Hi all,
For the DIAX users who use the CallMe feature in the Help menu, I kindly
ask you that if you call, to leave a message too. I have a lot of calls
using this feature and there is just a click. The user hangup without
leaving any message.
Thank you for your understanding,
Dan
quote who=Sathya Weerasooriya
I installed a x100P card today. Once it is configured * no longer
starting.
[snip]
[EMAIL PROTECTED] asterisk]# ztcfg -vv
Zaptel Configuration
==
Channel map:
0 channels configured.
Have you configured
Did you setup your zaptel.conf?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sathya Weerasooriya
Sent: Wednesday, November 05, 2003 2:14 AM
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-Users] X100P - module does not
Hello,
I am setting up 2 ISDN 4 linux cards and have had great success now that
I have got over the initial problems with : and / characters.
The only problem I am experiencing now is the sending of DTMF tones over
the line to a remote IVR system.
If I dial SIP (Cisco 7905 and 7940) to a
Hi,
Is it possible to develop a client (IAX/SIP/H323) and work from inside a browser (IE/NS/MG) for CRM reasons ? Any work around ? Suggestions ?
Is there a manager console over a browser ? If not, is there an intention to develop one ?
Also, is there a TCP/IP server to control * in order to
I have the MGCP Firmware and call transfer doesn't work in my configuration.
Daniel
Marian Danisek a écrit:
Daniel ANDRE wrote:
Hello,
Now that I have a nearly working configuration for my IP10S with * I
wonder if anyone has done call transfert with this Phone. In the
IP10S documentation
Daniel ANDRE wrote:
I have the MGCP Firmware and call transfer doesn't work in my
configuration.
show mgcp.conf
Daniel
Marian Danisek a écrit:
Daniel ANDRE wrote:
Hello,
Now that I have a nearly working configuration for my IP10S with * I
wonder if anyone has done call transfert with this
Hello Dan,
Its an excellent start. Please don't get swayed away by some stupid
remarks. I am really impressed by your work and I hope to see a lot more
releases from you.
In spirit of improving the code, here are some of the issues that I
faced while trying it out:
1. Once I dial the number,
I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI
(chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone
else has seen them:
Hmm, I'm running plain vanilla * v0.5 and have no problems with that
particular card, same version of chan_capi. Did you compile
Some more remarks:
1: Message: Unknown event: 6 for call 1
2. Message: No free call appearences ?
3. Again, two buttons grayed out?
ricky
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Wednesday, November 05, 2003 12:38 AM
To:
Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net
I've posted my demp weather report Asterisk AGI script at
http://www.fnords.org/~eric/asterisk/downloads/
Eric,
Can you comment on the difference in installation ease for Festival and
Cepstral?
Regards,
Robert
On Tue, Nov 04, 2003 at 07:24:19PM +0100, Olle E. Johansson wrote:
Keep feeding the list, I'll steel information to the wiki.
http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office
Superb stuff, Olle :)
If we can establish a 'standard format' (maybe an HTML form?) for
On Tue, Nov 04, 2003 at 01:52:46PM -0600, Steven Critchfield wrote:
- Has been in nearly fault free operation for more than since 05-2002.
Great stuff, Steven! :)
Can I enquire what was the cause of the downtime? Was it planned-
maintenance, or an actual fault with the Asterisk software /
In response to the SIP and NAT discussion, I have updated the ticket
on the subject that seemed to be getting the most attention: #104.
There are enough clueful people here that perhaps someone can come up
with a patch that handles NAT in the elegant way that I describe in
the bugnotes, as I
Hi,
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 10:55 AM
Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Some more remarks:
1: Message: Unknown event: 6 for call 1
This message is
It works with SIP and with zap channels.
What about IAX? like DIAX softphone?
I may be misunderstanding something.
When you start an Asterisk configuration process, connecting your hardware
and building your dialplan, you use Zapata.conf, sip.con and iax.conf to
connect the FXSs and FXOs to your
On Tue, 4 Nov 2003 14:06:43 -0800
John Todd [EMAIL PROTECTED] wrote:
Hi, * gurus,
I wonder if there is a way to alert the in-house extension(s) in case of an
incoming external call without actually answering it, before somebody picks
up the phone on one of the extensions? This way the
...and to solve another problem, there's my suggestion on support for outbound SIP
proxy.
http://bugs.digium.com/bug_view_page.php?bug_id=359
There are corporate networks that use a SIP proxy proxy as an ALG, application layer
gateway,
for all outbound and inbound SIP traffic in the DMZ.
Hi!
http://www.skype.com/
seesm to be the latest craze... anyone have any knowledge of their
technolgy use etc ??
- closed source
- WinXP and 2k only
- peer-2-peer, i.e. they route foreign calls through your client (and
bandwidth) if that helps the calling parties
Philipp
Philipp von Klitzing wrote:
http://www.skype.com/
seesm to be the latest craze... anyone have any knowledge of their
technolgy use etc ??
- closed source
- WinXP and 2k only
- peer-2-peer, i.e. they route foreign calls through your client (and
bandwidth) if that helps the calling parties
In
On 05/11/03 10:14, Olle E. Johansson wrote:
As I understand it they must not be fully peer-to-peer even if they
use your bandwidth, there has to be media servers in their network,
handling calls. Or?
No, the whole point is that it's completely decentralized. More
interesting to end users is that
Philipp von Klitzing wrote:
http://www.skype.com/
seesm to be the latest craze... anyone have any knowledge of their
technolgy use etc ??
- closed source
- WinXP and 2k only
- peer-2-peer, i.e. they route foreign calls through your client
(and bandwidth) if that helps the calling parties
In
On Wed, 05 Nov 2003 10:36:01 +, Alastair Maw wrote:
On 05/11/03 10:14, Olle E. Johansson wrote:
As I understand it they must not be fully peer-to-peer even if they
use your bandwidth, there has to be media servers in their network,
handling calls. Or?
No, the whole point is that it's
Alastair Maw wrote:
On 05/11/03 10:14, Olle E. Johansson wrote:
As I understand it they must not be fully peer-to-peer even if they
use your bandwidth, there has to be media servers in their network,
handling calls. Or?
No, the whole point is that it's completely decentralized. More
Peter Zeltins [EMAIL PROTECTED] said:
Hmm, I'm running plain vanilla * v0.5 and have no problems with that
particular card, same version of chan_capi. Did you compile fcpci driver
yourself? I'm on RH9.
Yes, I compiled it myself. I'm running on Debian unstable, kernel 2.4.21
(homebuild)
--
Cees
I just downloaded the newest version from CVS([EMAIL PROTECTED]) and I am getting an error whenever
I call the asterisk box. I cannot here any audio on the budgtone. This works
fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it
rings but I get that same error when I
Hi all,
i have purchased the g.729 codec from digium.
The registration was successful. (with the old binary)
But there're a few questions:
- should not the codec listed in the codec list when i enter show codecs
?
- the codec is named with g729b but if i enter show codecs there is a
Daniel ANDRE wrote:
I have the MGCP Firmware and call transfer doesn't work in my
configuration.
this is my mgcp.conf with working call transfer:
[general]
port = 2427
bindaddr = 192.168.1.253
[192.168.1.92]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
I am trying to write my first AGI script..
I cant seem to get it to work.. I am trying PHP in preference (I know
this is frowned upon) but I can't get it to work with perl either.. I
guess I just don't understand it correctly..
All I am trying to do is get the script to make a call using Dial
On Wed, 2003-11-05 at 04:55, Roy Sigurd Karlsbakk wrote:
Is there any information available about ZapRAS other than the fscking
source?
just pointing out for those interested in reading that I have written up
my recent experience over on the -dev list where I read this first.
Ollie, you want
Hi
?php
// From Kapjod's sample..
ob_implicit_flush(true);
set_time_limit(0);
$err = fopen(php://stderr,w);
$in = fopen(php://stdin,r);
$out = fopen(php://stdout,w);
//This works..
fputs($out, Verbose \Calling phone\n);
// This doesn't
fputs($out, exec(Dial(sip/2012)\n);
fclose($in);
On Wed, 2003-11-05 at 03:11, Gavin Hamill wrote:
On Tue, Nov 04, 2003 at 01:52:46PM -0600, Steven Critchfield wrote:
- Has been in nearly fault free operation for more than since 05-2002.
Great stuff, Steven! :)
Can I enquire what was the cause of the downtime? Was it planned-
On Wed, 2003-11-05 at 02:32, Shoval Tom wrote:
It works with SIP and with zap channels.
What about IAX? like DIAX softphone?
I may be misunderstanding something.
When you start an Asterisk configuration process, connecting your hardware
and building your dialplan, you use Zapata.conf,
Hi all,
Solution found.
Asterisk CVS-10/29/03 is simply BAD for chan_capi and
X-ten Lite.
New version Asterisk CVS-11/04/03 does the job.
Haven't done much testing but 2 way sound is there.
So everybody using this version of asterisk go for a
new cvs...
Thanks,
Thorsten
I want to setup an Asterisk network using
the Cisco ATA 186.
What is the best place to order those devices?
I'm not finding them anywhere.
Al
__
Do you Yahoo!?
Protect your identity with Yahoo! Mail AddressGuard
http://antispam.yahoo.com/whatsnewfree
Paul Liew wrote:
Hi
?php
// From Kapjod's sample..
ob_implicit_flush(true);
set_time_limit(0);
$err = fopen(php://stderr,w);
$in = fopen(php://stdin,r);
$out = fopen(php://stdout,w);
//This works..
fputs($out, Verbose \Calling phone\n);
// This doesn't
fputs($out, exec(Dial(sip/2012)\n);
Hi,
- Original Message -
From: Shoval Tom [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 1:34 PM
Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Dan, I can't seem to transfer calls using #.
How is it supposed to be done?
In the
On Wed, 2003-11-05 at 12:36, Al wrote:
I want to setup an Asterisk network using
the Cisco ATA 186.
What is the best place to order those devices?
I can't speak for the best place... but I found this:
http://www.pricegrabber.com/search_getprod.php/masterid=614321
However, given that this
Hello there,
Does anyone know how an agent logged on with AgentCallbackLogin
application can logoff ?
Thanks,
Anna
Anna Panagidou
Technology Department
Hellas On Line
Agiou Konstantinou 59-61
15124, Maroussi
Tel. no: (+30210) 8762309
E-mail address: [EMAIL PROTECTED]
Best of luck on getting an answer, I have posted several times with the
same question.
Unfortunately my time to reverse engineer this problem right now is low
but my
temporary solution's cons are pushing me to jump into the code and fix
the problem.
As a workaround you can set your Cisco phones
Hi i'am again...
i have tesed if my * (where the purch. g729 is installed) take calls from a
gateway with g.729A codec.
The calling mechanism works but there is no voice only bad noises .
I'am a little bit confused.
On the digium site i bought a g729 codec (without any indication of an a
Steven Critchfield wrote:
On Wed, 2003-11-05 at 04:55, Roy Sigurd Karlsbakk wrote:
Is there any information available about ZapRAS other than the fscking
source?
just pointing out for those interested in reading that I have written up
my recent experience over on the -dev list where I read
Hi,
Is there any (easy) way to get Asterisk to include CIC-information in
the SIP INVITE?
CIC:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/chapter3.htm#1314580
I need my SIP INVITE to look something like:
INVITE sip:5550001;[EMAIL
I wonder what would be the easiest way to con
Asterisk into logging all activityon ISDN line? Likeincoming calls,
outgoingetc, even if these calls did not originate/terminate at Asterisk
server? I'm using chan_capi if that matters (it should), with Fritz PCI
S-type ISDN connection.
TIA,
I've had t a the end, I've added T, although this is not necessary.
Still doesn't work, though
When calling or called party press #, nothing happens. Asterisk's console
doesn't show anything, either.
Can you send me a sample of an extension definition that works?
-Original Message-
Hi,
- Original Message -
From: Shoval Tom [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 2:32 PM
Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
I've had t a the end, I've added T, although this is not necessary.
Still doesn't
At 08:34 PM 11/4/2003 -0600, you wrote:
Quoting Chris Ziomkowski [EMAIL PROTECTED]:
I can't try this setup yet (still don't have the hardware), and have been
trying to answer this question merely from the source code. So far, I have
not been able to convince myself. Does anyone have definitive
Brian,
I think some of the confusion comes from what end of the line we are
looking at and the nature of the imbalance. While the resistor may fix
the near end, it will probably cause some termination problems at the
far end. Reflections mostly, which on a short run analog line shouldn't
be
-- Original Message --
From: Dan [EMAIL PROTECTED]
Hi,
Hi guys
As usual I am playing around with IAX soft clients. I was wondering with
the various IAX clients, IAX client, DIAX, etc how's one park calls,
transfer calls, etc since there is no flash
Hello all,
A newer version of asterisk-oh323 is available. This version
features a set of channel variables and improvements in audio frame
handling. People that have reported clicks or choppy sound, in
some cases, should try this version.
Download from:
Hi,
..
But due to allot of systems needing the # we have disabled it here in our
Asterisk system. The # key is used for pagers and other calls. That is why
we would like to get a flash key!
I don't think is implemented in the library..:(
By the way the sound is better with DIAX then with
Hello Al,
Please let me know how many of them do you want to order, we are
distributors of Cisco and can help you . Als owe are providing
International/Domestic calls termination to more then 260 countries
worldwide.
Thanks,
Alexander
Unofficial Asterisk Forums
Jorge Mendoza wrote:
I'm in agree with all explanations regarding the echo and 2/4 wires
conversion. However I'm wondering if there are other parameters like
CPU and/or Asterisk configuration involved in the problem with more
weight than hybrid. Otherwise how do you explain the difference in
Has anyone used Gastman/Astman successfully?
I have it up and running (Gastman win32), but have a problem with the
creation of end stations on the map. I'm not sure of the format of the
extension to use when creating a end station icon.
Services like Conference bridge and Musichonhold seem to
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of is 'Relatively few installations
worldwide'
Can
Lee Goodman wrote:
Has anyone used Gastman/Astman successfully?
I have it up and running (Gastman win32), but have a problem with the
creation of end stations on the map. I'm not sure of the format of the
extension to use when creating a end station icon.
Services like Conference bridge and
On Wed, 2003-11-05 at 09:08, Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of
Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of is 'Relatively few
On Wed, 2003-11-05 at 08:32, Dan wrote:
Hi,
..
But due to allot of systems needing the # we have disabled it here in our
Asterisk system. The # key is used for pagers and other calls. That is why
we would like to get a flash key!
I don't think is implemented in the library..:(
Hi Everybody
I'm connecting a zplex 10B to a TE410P card on my asterisk server (i use a
cross cable), the zplex doesn't show any alarm neither the server. I'm
trying to make calls from 1 line from the channel bank, to another, but when
I dial the ext number, the dialtone doesn't stop... every
Michael Manousos wrote:
A newer version of asterisk-oh323 is available. This version
features a set of channel variables and improvements in audio frame
handling. People that have reported clicks or choppy sound, in
some cases, should try this version.
Download from:
Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only Con I can think of is 'Relatively few
Is
it possible to use * as a VOIP gateway?
Can
I connect asterisk to one of the trunks on my current PBX and on the other side
of the world connect another * to the trunk of another regular PBX is it
possible to transfer calls from here to there?
I
guess I'll need one port FXO card
On Wed, 2003-11-05 at 15:31, Steven Critchfield wrote:
the only Con I can think of is 'Relatively few installations
worldwide'
Your listed con is only a con if you have to point to others failures to
cover your own. It is lemming thinking.
Hm, it's only a con because I can't think of
On Wed, 2003-11-05 at 09:36, WipeOut wrote:
Gavin Hamill wrote:
It would seem an odd question, but I'm trying to put together a little
presentation on 'Why Asterisk?' and need to list Pros and Cons I've
got plenty of Pros (including the availability of commercial support),
but the only
Cepestral was installed and working within 10 mins of my decision to
purchase it. It's $30.00 and can be purchased on their web site and
they give you a download. They have a demp on their website that will
do text-to-speech and give you a .wav file to download and listen to.
Download,
Hello,
Can somebody point me where I can find examples how
I can set up Remote Call Pickup ?
Thx
Bart
On Wed, 05 Nov 2003 09:36:28 -0600, Steven Critchfield wrote:
On Wed, 2003-11-05 at 08:32, Dan wrote:
Hi,
..
But due to allot of systems needing the # we have disabled it here in our
Asterisk system. The # key is used for pagers and other calls. That is why
we would like to get a flash
One of the biggest cons is the lack of friendly interface for
configuration. However, most PBXs in use don't have one either, unless
they are about 5 years old or newer, in which case it probably wouldn't
be on the chopping block.
I still think the pros outweigh the cons, or else I wouldn't be
On Wed, 2003-11-05 at 16:02, Olle E. Johansson wrote:
Cons:
* Not a full SIP proxy
Fortunately this is not relevant to our environment :)
* No release handling
Good point, I've added that to the list..
* Limited hardware support
The software is pretty well tied to Digium hardware for
Olle E. Johansson wrote:
Michael Manousos wrote:
A newer version of asterisk-oh323 is available. This version
features a set of channel variables and improvements in audio frame
handling. People that have reported clicks or choppy sound, in
some cases, should try this version.
Download from:
Hello,
I have received yet another new phone today, the ClipComm 101
(http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html)
I bought it for $165 directly from the Korean Manufacturer(No US distributer
yet). Here are the features:
- Built-in NAT functionality, you can switch from
Thomas Haeger wrote:
Hi all (Michael),
how it is possible to get the ip address of the calling party ?
(i know by using h323... but there're a few unknown segfaults...) and so i
want to use oh323, but i have to get the ip from the caller to permit or
deny the call with AGI.
Is it possible at all
Hi,
Sorry to answer just now. your mail was lost between others from the
same list..;)
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 10:37 AM
Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
yes you can but may not be all that economical though.
on the other hand, if you can replace or do away with
at least one of the pbx with * at either end,
i think you'll be ahead of the game :-)
Shoval Tomer wrote:
Is it possible to use * as a VOIP gateway?
Can I connect asterisk to one of the
--- Gavin Hamill [EMAIL PROTECTED] wrote:
It would seem an odd question, but I'm trying to put together a
little
presentation on 'Why Asterisk?' and need to list Pros and Cons
I've
got plenty of Pros (including the availability of commercial
support),
but the only Con I can think of is
On Wed, 2003-11-05 at 10:16, Gavin Hamill wrote:
On Wed, 2003-11-05 at 15:31, Steven Critchfield wrote:
the only Con I can think of is 'Relatively few installations
worldwide'
Your listed con is only a con if you have to point to others failures to
cover your own. It is lemming
Hi Steven,
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 5:36 PM
Subject: Re: [Asterisk-Users] IAX clients and the flash button
...
Dan, Ariel is asking you to attack this problem incorrectly. You are
correct
Hi Andrew,
Tuesday, November 4, 2003, 5:04:04 AM, you wrote:
AJ I have used G723.1 (although unlicensed) with Asterisk. The info is even
AJ in the Makefile, just drop in a few files in your source directoy,
AJ uncomment something in the Makefile and instant G723.1 support...
Thanks for the
On Wednesday 05 November 2003 08:47, Lee Goodman wrote:
Has anyone used Gastman/Astman successfully?
I have it up and running (Gastman win32), but have a problem with
the creation of end stations on the map. I'm not sure of the format
of the extension to use when creating a end station icon.
Yes, I agree. Your typical PC might have only 96% uptime
but you could still build a __system__ with five nines of
uptime using PC hardware.
You eed to do two things. 1) Use better quality PC hardware
that employs some internal redundancy, like mirrored drives and
multiple load sharing power
i m a newbie with * so in all likelihood my question will sound stupid
to you but aren't there HA support for linux already?
as to the pstn interfaces, i thought most traditional PBX uses redundant
equipment to provide HA;
can't we do the same with * being the switch?
WipeOut wrote:
Gavin
Steven Critchfield wrote:
I think the number you cited needs qualification to be accurate. Because
if it where accurate as it stands, I'm due for major downtime in my rack
as I have several systems approaching 2 years uptime without a single
hardware failure. These machines also where not new
Hi all,
I have done some minutes ago a full CVS update, like that:
cvs checkout zaptel zapata libpri asterisk
cd zaptel
make clean ; make install
cd ../zapata
make clean ; make install
cd ../libpri
make clean ; make install
cd ../asterisk
make clean ; make install
When I try to start astersik
I did cvs update on asterisk, zaptel, libpri as of today (November 5,
2003). I also did 'make clean' on each of them. My previous version of
asterisk was cvs of September 15, 2003. No other changes have been made
to my system other that these updates.
when running
'make asterisk'
the following
Greetings to all,
We are new to the Asterisk community and really
appreciate thewillingness you have to share knowledge and help one
another. We are building one of the Nations first fiber to the home networks
here in Provo and have selected Asterisk to test as our phone
switch.
Newbie
I have a cisco 7910 phone, I'm trying to get it to connect to asterisk,
But it seems like it needs either a SEPDefault.cnf file or a
SEPMACADDR.cnf file to
Continue, I created empty ones but it's still sitting there saying
opening
Does anyone have examples of the SEPDefault.cnf file?
Kevin,
hello!
I have active call from i4l modem to ZAP (FXS).When someone on i4l
(telco side) speaks i hear DTMF tones on other side (ZAP).
How to turn off DTMF detection on modem-i4l side ?
Is it possible to do that ??
status of active channels:
server*CLI show channel Modem[i4l]/ttyi0
-- General
Can anyone think of any others?
mmh... some idea here
* experienced linux user for production use
(able to di compilation, knows how the shell works,
able to debug code kernel probs, blah blah blah)
* interoperating with other telco (even only lines...)
needs some background in telecom
Since it's all the craze, I might as well post our current Asterisk
usage. :-)
EQUIPMENT:
- Beefyish box (dual Xeon 2.4GHz, gig of RAM, more-than-adequate disk
space, etc) in a 1U chassis.
- A second, slightly less beefyish box of specs I don't have handy right
now, also in a 1U.
-
Hello all--
When I bought the TDM400P and the two FXO cards to prototype (small-scale) what could be done with Asterisk, I got a single sheet of paper with the cards, that explained how to insert the card, and fetch the source for asterisk, zaptel, and whatnot. But, before I could get it
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has
gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway
working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) --
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has
gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway
working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) --
On Wednesday 05 November 2003 12:03, Dan wrote:
[app_voicemail2.so]WARNING[1074412256]: File loader.c, Line 232
(ast_load_resource): /usr/lib/asterisk/modules/app_voicemail2.so:
undefined symbol: ast_localtime
WARNING[1074412256]: File loader.c, Line 400 (load_modules):
Loading module
I just got mine working. All I did was create a skinny.conf and point the
phone to the asterisk server for tftp. the phone then boots and says useing
TFTP as CM and works. I have no SEP.cnf's on my tftp server. my skinny.conf
is
[general]
dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
This company seems to think pros outweigh the cons for Asterisk:
www.voicepulse.com
/. reported today that VoicePulse uses a variation of Asterisk to run
their Broadband Phone Service.
http://slashdot.org/article.pl?sid=03/11/05/1319251mode=threadtid=126
Steven Critchfield wrote:
On Wed,
Hello.
I have a problem
I
want to pass the archives of the voicemail of the a Spanish
They
can say to me that software I can use to create the archives gsm ..
J.R
It is in fact G729A
User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format
10 00070ea6-2f 00101/00103 0ms ms G729A
1 active SIP channel(s)
Thanks,
Brian
On Wed, 5 Nov 2003, Thomas Haeger wrote:
Hi i'am again...
i have tesed if my * (where the purch. g729 is
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