Make this another vote for Zap and IAX2 monitoring :)
Paul
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 17, 2005 12:10 AM
Subject: SV: SV:
There was some talk last June about some folks trying GR303 with *.
Asterisk supports GR-303 access concentrators now; I do not know if the
support is in stable, or only in CVS HEAD.
Asterisk does not know how to act _as_ an access concentrator, however.
Do you have an recomendations for
Hi,
I believe there is a script that reads the contents of voicemail.conf and
goes on to send the voice e-mail messages to whatever e-mail address
specified in voicemail.conf. What's the name of this script and where is
it located?
Thanks,
Julius.
Hello,
I tried to connect my Siemens ISDN 4170 cordless small PBX to the
asterisk with an Acer HFC isdn card but I have not seen any success. The
asterisk can see the card but no success to receive or dial a call. The
Siemens 4170 gives fault in all the handsets with trying to dial and
does
Setup:
POTS phone1 - Panasonic Analog PBX - Digium FXO - Asterisk1 - IAX2 -
Asterisk2 - Digium FXS - POTS phone2
I am attempting to balance the Digium FXO (shown above), analog audio
levels using ztmonitor -v, which the information I have found means
getting the TX and RX indicators to hit
From the Asterisk 1.0.7 changelog:
Asterisk 1.0.7
-- chan_skinny
-- A check has been added to avoid a crash.
Will this fix be ported to chan_sccp too? chan_sccp still bombs out
regularly (segfault)
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i have been using the asterisk for some three
weeks. Previously i was using the softphone iax-phone and now i have to shift to
the sip phone xlite.
The problem is that there's always unbearable noice
in sip to sip calls. Is there any way to get rid of this
Kindest
MM Luqman
How can I upgrade
Asterisk to the latest version ??
Will I need to
re-compile??
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hello,
can anyone installing/configuring asterisk's on SLES9 if someone can
share his/her views experiences .
Thanks In Advance.
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To
On Thu, Mar 17, 2005 at 12:02:43PM +0100, Kib Eki wrote:
I found the chan_capi for asterisk from www.junghanns.net. Also loaded
the patch and applied to the chan_capi source tree.
The patch - is it this one?
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
I changed the
Hi Kong,
IAX2 support has been added in release 0.65 of IPSwitchBoard, which is ready
for download at
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA, Zap is next
(I just need to get a Zap card).
Thorben
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED]
On Sat, Mar 19, 2005 at 10:40:10AM +0100, Reuben Grech wrote:
How can I upgrade Asterisk to the latest version ??
You will have to use cvs:
http://www.automated.it/guidetoasterisk.htm#_Toc49248761
Will I need to re-compile??
Yes.
-Thomas
___
On Fri, Mar 18, 2005 at 12:14:14AM +0800, Craig Guy wrote:
Upgrade to kernel 2.6.9, there are supposed to be significant bugfixes for
CAPI support in 2.6.9.
All of my CAPI systems use FC2, 2.6.9. I tried to go 2.6.10 but had
problems.
What kind of problems. Could you give us some hint?
Hi,
I have a server with 2 TE110P cards. 1 card is plugged to telco line,
another card is plugged with a Hicom PBX.
I want to send some call to VoIP phones and all other to my PBX.
I don't known how to make my dialplan :
===Extensions.conf==
[incoming_call]
exten =
Hi friends,
i`m totally a newbie on VoIP let alone asterisk.
I`m very much interested in learning asterisk to deploy on my small
Call Center, we have 2 audioCodec MP-108 8 fxs port SIP device and 6
A800 H323 analog quintums.
I installed fresh asterisk with samples, might b peice of cake for u
can you force polycom hones to handle multiple lines
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 17, 2005 3:06 PM
Subject: Re: [Asterisk-Users] Polycom vs. Cisco IP
On Sat, 19 Mar 2005, Jeremy SALMON wrote:
I have a server with 2 TE110P cards. 1 card is plugged to telco line,
another card is plugged with a Hicom PBX.
As a side note (not related to the problem at hand) using two TE110P cards
is really suboptimal since the clocking is not passed between
I'm having with an echo or delay
I connect to the PSTN with a x100p and then connect a std. phone
to a FXS module on a TDM10B.
The std phone is only 2-wire so I know this is not helping.
(yes I have read the 2-wire 4-wire issue)
I have tried many echocancel values. The best thing to
Hi,
I'm trying to install Areskicc but with no much sucess. After
installing the UI I login as root/mypass, but in the left menu I only
have 3 options: Main Template, disconnect and logout. When I click on
Mail Template Show mail template it shows an error page. First, is
there a way to increase
On Sat, Mar 19, 2005 at 07:19:39AM -0600, Rich Adamson wrote:
If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.
If you have a x100p clone, it is likely the problem. Replace it with
Hello,
We created an areacode, country code, GMT offset, country code file for the
astGUIclient project last year. I believe it has all Mexican area codes in
it. If you find any errors we've love to hear about it.
http://astguiclient.sourceforge.net/phone_codes_GMT.txt
Hope this helps,
MATT---
mattf wrote:
Hello,
We created an areacode, country code, GMT offset, country code file for the
astGUIclient project last year. I believe it has all Mexican area codes in
it. If you find any errors we've love to hear about it.
http://astguiclient.sourceforge.net/phone_codes_GMT.txt
Hope this
I bought a couple Polycom Soundpoint 300's, and have them working nicely
with SIP... but I'd like to be able to do automatic config via FTP, but it
requires some XML config files. The docs discuss them in detail, but I
can't seem to d/l them from Polycom. [No, it doesn't appear to be on
Hello. I would like to know if somebody did a wireles voip with Asterisk PBX.
I think to deploy a wireless for about 500 potential customers, it's a 3 km
radius maximum coverage with houses without phone lines, I work for public
places telephony small enterprises ( a common bussines in
/etc/php4/apache/php.ini
register_globals = On
solved my problem.
On Sat, 19 Mar 2005 14:26:27 +0100, Tacio Santos [EMAIL PROTECTED] wrote:
Hi,
I'm trying to install Areskicc but with no much sucess. After
installing the UI I login as root/mypass, but in the left menu I only
have 3 options:
A lot of the BV config confusion is the result of users with registered
IP's vs nat'ed IPs. The patch _was_ only required for those that used
nat'ed systems (proven shortly after that patch was released, and backed
by those that wrote the patch).
So, for those that are still mucking around with
Hi Thorben,
I have just installed version 0.65. When I click on an extension to dial it dials the number that is the extension of the monitor. The weird thing is that in the * console the dial command shows the actual extension intented to be dialed (but it is not).
Max.
Op za, 19-03-2005
I finished my setup for ASTCC and I am looking for a tool to convert a
mysql table to excel and back.
Which one is good, and where can I find it?
bye
Ronald
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Thanks for the help. That was it!
Marios Andreou wrote:
Hello Justin,
dtmfmode should be inband for broadvoice either way because that's what they
support.
Now for the extensions.conf do you have:
exten = ,1,Dial([SIP|IAX2|..]/something, timeout, t) -- t for
transfers?
-Original
If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.
If you have a x100p clone, it is likely the problem. Replace it with
something capable of matching the pstn impedance for whatever
I have tdm400p with 4 fxo modules on it. When I call into the asterisk
box from my mobile, I can see the asterisk console picks the call up
and routes it to my computer with x-lite. There was no sound coming
from either - just silence. I then decided to route it directly to
voice mail to
On Mar 18, 2005, at 10:47 PM, Russell Bryant wrote:
Hello everyone,
Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now
been released. Libpri and -addons have not changed, but have been
updated anyway to keep the version numbers consistent. All of the
tarballs are available
hi
i wonder why my outbound calls via asterisk-sipgate-german telecom have such
high delay rates (about 500 or mor ms) while inbound signals are quite
ok (max ca 200ms). any idea?
joerg
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I got an error when compiling asterisk 1.0.6
res_crypto.c: En la función `crypto_init':
res_crypto.c:553: aviso: declaración implícita de la función
`SSL_library_init'
res_crypto.c:554: aviso: declaración implícita de la función
`ERR_load_crypto_strings'
make[1]: *** [res_crypto.o] Error 1
* Pol ([EMAIL PROTECTED]) ha scritto:
I got an error when compiling asterisk 1.0.6
res_crypto.c: En la función `crypto_init':
res_crypto.c:553: aviso: declaración implícita de la función
`SSL_library_init'
res_crypto.c:554: aviso: declaración implícita de la función
Hi,
I am still looking for confirmation that ChanISAvail in CVS current doesn't
work properly anymore.
My config hasn't changed (it worked for months)...
Right now, every time ChanIsAvail jumps to n+101 regardless if tested
channel is available or not.
Is it broken? Maybe the syntax has changed?
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
Also, is it possible to put a phone into multiple contexts?
For instance:
Anyone successfully implemented a solution for allowing ZapBarge call
monitoring only for a specific group of agents calls?
The issue I see is that the feature only works on zap channels, and all
of the agents (in many cases) are IP phones.
Allowing ZapBarge and ZapScan on the TDM PSTN (t100p)
Good afternoon
Since the cvs version of yesterday, the ip address and the port of the
sipfriend are no more updated in the realtime database
Regards
Thierry Wehr
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How can I change the IRQ of the cards?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Viernes, 18 de Marzo de 2005 12:15 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voice getting
Looking at that list, the easiest way would be to disable all your USB
ports in your BIOS, reboot and see if the card has its own IRQ. Assuming
you don't need USB. In general, just turn off all the things you don't
need that use IRQs.
-Original Message-
From: [EMAIL PROTECTED]
On Sat, Mar 19, 2005 at 09:22:22AM -0600, Rich Adamson wrote:
If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.
If you have a x100p clone, it is likely the problem. Replace it
On Thu, 17 Mar 2005, Christopher Jacob wrote:
[Deleted]
So, the moral of this story
While Polycom may not offer configuration type support for asterisk, they
stand by their hardware. With Cisco you have to shop around to find a decent
deal, and who know how you're going to get support.
You really need an outside company to implement e911 for you. Your idea
of putting each phone in a separate context might work if;
You have a local emergency number for every caller in your dial plan.
You have maps that show the jurisdiction of every 911 dispatcher in your
service area.
You
You should have set up the two cards as zaptel as a different group in
the zapata.conf.
Then if you want to dial your pbx you are dialing out of Asterisk, so
you use the Dial command.
Assuming that the PBX PRI link is in group 2 in zapata.conf
Something like:
exten =
Matt Darnell wrote:
Do you have an recomendations for the GR-303 concentrator?
There are a number of them; Digium has an Adtran unit (IIRC) that they
demonstrate in their trade show booth.
I was read that the GR-303 protocol is very similar to ISDN-PRI NFAS.
Yes it is.
I don't understand what
Does anyone know what a sample extensions.conf and/or sip.conf would
look like to place a call on mute? Also, what does a sample
extensions.conf file look like for three way calling.
Thank you,
Justin Ramsey
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Well.. I dont use USB nor ps2 mouse ports so.. Lets try that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Scott
Sent: Sábado, 19 de Marzo de 2005 10:12 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
If you have any experience using * (or VoIP in general) with DirecWay,
please respond privately. I am particularly interested in experiences in
Latin America.
TIA!
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.
If you have a x100p clone, it is likely the problem. Replace it with
something capable of matching the pstn impedance for
[EMAIL PROTECTED] is believed to have said:
Hi Kong,
IAX2 support has been added in release 0.65 of IPSwitchBoard, which is ready
for download at
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA, Zap is next
(I just need to get a Zap card).
Thorben
Hi Thorben,
IPSwitchBoard
--On Saturday, March 19, 2005 8:45 AM -0800 Bruce Komito [EMAIL PROTECTED]
wrote:
If you have any experience using * (or VoIP in general) with DirecWay,
please respond privately. I am particularly interested in experiences in
Latin America.
TIA!
I have an Asterisk system on the net, and I
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if they could produce one for
say $1000 retail?
Hi,
In the current asterisk release 1.0.6 does G726 currently support
16/24/32/40, or are we still only at 32kbps?
Are there any plans to allow variable packetization rates per sip
device for applications like faxing over voip, etc?
___
Asterisk-Users
Rob Scott wrote:
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Don't be surprised if you see something like that soon.
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How to install /use festival on Asterisk?
I would need text to speech in:
English
German
andChinese (Mandarin)
bye
Ronald
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To
It seems to me silly to have a T1/E1 card to connect to a
channel bank when you could just have a 24/30 way FXS card in the
slot in the first place.
Wouldn't a SIP channel bank be better - something that has multiple
FXS and FXO ports but hooks up to Ethernet. I know Wasam (ala Farfon) is
try
I would suggest contacting a dealer until you find one who will sell
you a maintenance contract for the phone. Last I checked, over a year
ago, they were somewhere around $10-$20. Once you have a contract you
may register online and download all the software you need. However to
use legally
Jerry wrote:
I would suggest contacting a dealer until you find one who will sell you
a maintenance contract for the phone. Last I checked, over a year ago,
they were somewhere around $10-$20. Once you have a contract you may
register online and download all the software you need. However to
I wonder if VoipSupply can sell the maintenance contract for the phone?
Wouldn't hurt to ask. The fellow from VS is a regular poster over on the
asterisk-biz list.
--On Saturday, March 19, 2005 11:09 AM -0600 Jerry [EMAIL PROTECTED]
wrote:
I would suggest contacting a dealer until you find
Matt wrote:
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
I think that's what you'd have to do.
Also, is it possible to put a phone
I can't figure out how one upgrades the boot ROMs for the Polycoms.
I've read the wiki, I've checked the docs, I've looked here, there,
everywhere; I hoped it might be through the web interface, through
DHCP... but I can't see how to do it.
Suggestions?
Thanks!
-Ken
In article [EMAIL PROTECTED],
J Thomas [EMAIL PROTECTED] wrote:
I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I
consistently get one of the following errors:
PRI got event: HDLC Abort (6) on Primary D-channel of span 1
or
PRI got event: HDLC Bad FCS (8) on Primary
Good Day list,
Need assistance determining the best place to read up on whether
Asterisk can help me out.
I have a situation where I need to do the following
PRI from Telco ---
Analog Channel BankProprietary Box
Just use the authenticate app.
show application authenticate
Damon Estep wrote:
Anyone successfully implemented a solution for allowing ZapBarge call
monitoring only for a specific group of agents calls?
The issue I see is that the feature only works on zap channels, and all
of the agents (in many
Ken D'Ambrosio wrote:
I can't figure out how one upgrades the boot ROMs for the Polycoms.
I've read the wiki, I've checked the docs, I've looked here, there,
everywhere; I hoped it might be through the web interface, through
DHCP... but I can't see how to do it.
Put the new bootrom.ld and
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing
done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
I think that's what you'd have to do.
Be careful. 911 centers are not
I have Meetme2 (as well as the web ui) installed and am having some
difficulty with the admin features.
I've set up two extensions pointing to the same conference, one with the
admin flag (1234|Maps) and another with (1234|Mmps). My issues:
If the admin presses the * key, it goes to an endless
I?m a telecommunication engineering student. I?m working on my degree thesis,
it?s about Astrerisk . My goal is to estimate the performance of a hybrid
platform for the Volp.
I?m looking for documentation about:
? Architecture
? Tools for the performances? analysis (to analyse
My Cisco 7960G is no longer a killer doorstop! It's now a fully
functioning phone! Very cool. Thanks to all.
Turns out that the 7.X series firmware does include a .loads file. It
looks to be a loader file used by Cisco's Application Manager to
identify and load the firmware version.
---
John
Do a search on Empirix.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, March 19, 2005 1:39 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk
I?m a telecommunication engineering
There was a big discussions at the Future of IP Heavyreading.com
conference in NY this week about this.
It's not as easy for the ISP's to get away with it as you think.
Empirix.com were called in to investigate and document the recent Vonage
case (lol interestingly enough they have also been
After looking everywhere I still don't have any solution. Transcoding
G729 is not a problem, Digium is selling licences at reasonable price,
but transcoding G723 is a huge problem (look at the prices!). And the
fact is that this is a quite often used codec.
I receive G729 G723 calls that I
When I use DISA I get congestion when I try to reach 1-800-number:
Here is the context:
[disa]
exten = 087,1,Answer
exten = 087,2,DigitTimeout,8
exten = 087,3,ResponseTimeout,20
exten = 087,4,Authenticate(985)
exten = 087,5,DISA(951|disa-access)
[disa-access]
include = tollfree
include =
On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote:
Jose R. Ortiz Ubarri wrote:
Hi:
I had asterisk with RealTime database working perfectly in a RH 9.0
machine. I used the sip cache so I even had MWI working. The problem
is that I decided to move to Fedora Core 3. I installed the
I'm ex cisco (left in '96). Cisco did change their corp policy recently
in that they no longer will sell firmware directly to end users. I think
as far as the phones go, that's a mistake on cisco's part. Perhaps we'll
see cisco move some of these phones over to the linksys side someday.
Any
PhpMyAdmin can export to excel.
-Matthew
From: Ronald Wiplinger [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sat, 19 Mar 2005 23:19:15 +0800
To: Asterisk Users Mailing List - Non-Commercial Discussion
+++ James Coberly [18/03/05 10:28 -0700]:
Try DIAX.
http://www.laser.com/dante/
Didnt work for me I logged into asterisk as an agent from diax but it didnt
open up a browser for the url i set in the queue options :(
Vikram Rangnekar wrote:
I have googled for days abt this so finally i
Hi,
I am having problems with callerid name and the time with my
dvg-1120S. Every time I receive a call, it reverts the phone to
January 1st 12:00am. I've looked everywhere in the browser and telnet
configuration to change this.
Also, it never shows the name of the caller. I've even tried forcing
I have a strange problem with my Polycom IP300 and Asterisk.
When I call the phone, the callerid is presented normally.
But when I go into the phones missed call list and try to call back
one of the numbers, the phone calls its own number no matter who the
caller was!
Has anyone seen this
I have the same problem but not with X-Lite. I was using Broadvoice
all day today and then I changed rate plans because I thought
everything was working well. Now my calls get dropped within 2 minutes
and my incoming calls go direct to broadvoice voicemail.
MARK.
Scott Wolfe wrote:
Hi,
- Original Message -
From: Vikram Rangnekar [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 19, 2005 9:30 PM
Subject: [Asterisk-Users] Re: Optional URL in App. Queue
+++ James Coberly [18/03/05
On Thu, Mar 17, 2005 at 08:33:28AM -0500, Ye Li wrote:
Hi there,
Newbie questions on ZAP channel numbering (forgive me if this was asked
before):
1. How are channels numbered if I have multiple FXS/FXO cards in the
system? Is there a fixed mapping between PCI slot id and the number
I think you're looking for the 'ChanSpy' application that seems to have
inexplicably vanished from the asterisk CVS..
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanSpy
If anyone has any info on this, let me know as I'm in a similar
situation.
Thanks,
tf.
On Sat, 2005-03-19
Hi,
If I have a PRI card in my asterisk server and have VoIP dial-tone
from Level3 over Ethernet, how do I go about setting up a routing
table to route all calls out over level3 with the exception of:
Any calls which would be local on the PRI (certain exchanges which I
will program in), and
On Fri, Mar 18, 2005 at 02:57:11PM +0100, Frank Fischer wrote:
Hi all
i'm just starting to setup my own asterisk. My first question is, if there
is any reason to choose a special linux distribution or if it doesn't mater
which distribution i chosse. Is there anything i should be aware of?
On Sat, 19 Mar 2005, Ronald Wiplinger wrote:
Russell Bryant wrote:
Hello everyone,
Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now
been released. Libpri and -addons have not changed, but have been
updated anyway to keep the version numbers consistent. All
On Sat, 19 Mar 2005, Paul Fielding wrote:
Make this another vote for Zap and IAX2 monitoring :)
Paul
Seconded! As cool as FOP is, this looks pretty awesome!
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It's been down the last 5 hours at least. Anyone know what the problem is,
or when it will be back up?
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off-topic
On Fri, Mar 18, 2005 at 11:56:19AM -0500, Alexander Lopez wrote:
On Friday, March 18, 2005 11:15 AM, Kanishka Somaratne wrote:
how do i reply a question asked in this mailling list.
I think you just did
Actually, he did not. He did the proper thing and posted a new message.
that seems to have done it!! _ALERT_INFO works great for the current
CVS, although it didn't seem to work in 1.0.5.. Thanks for the help!!
On Sat, 19 Mar 2005 16:52:26 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
You could try _ALERT_INFO instead of ALERT_INFO...
--
Cheers,
Matt
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
should tell u everything u need.
tf.
On Sat, 2005-03-19 at 15:51, Matt wrote:
Hi,
If I have a PRI card in my asterisk server and have VoIP dial-tone
from Level3 over Ethernet, how do I go about setting up a routing
table to
On Sat, 19 Mar 2005, Scott Gruby wrote:
On Mar 18, 2005, at 10:47 PM, Russell Bryant wrote:
Hello everyone,
Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now
been released. Libpri and -addons have not changed, but have been
updated anyway to keep the version
Like any other Distro war, it's all a matter of personal taste. For me,
I prefer gentoo (although I never have any luck with the asterisk ebuild
and end up building myself.. but typically, the package managment is
great).
If you have 0 linux experience, there are some distros that are much
i would recommend renaming or deleting /usr/lib/asterisk/modules, i
beat my head on the wall for an hour or so with this when upgrading
asterisk and trying to downrev back to 1.0.5 when i was having
problems with the latest cvs (which turned out to be a simple config
mod). so if you upgrade and
I think first we would need to clarify the use of the term FXO in this
context. I'm not a telco expert by any stretch, but it appears to me the
term is used misleadingly sometimes. It seems to me that an FXO port is a
port that you can plug an external phone line into - it can then allow you
Guys.
Anybody usign mysql addons for cdr? I just noticed that records are still
been sent to the master.csv file but seems not all of them.. Which makes me
think about which one has the actual truth? And why are not all records been
sent to the csv files and mysql?
Also, can I then disable the
I guess the first time it didn't get through... I didn't see it appear
in the list, that is...
I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on
it. They seem to run fine inside my network, so that's OK.
Now, I want to start using a X100P to connect it to my phone
Just wondering if there are any Zaurus owners out there using there
zaurus as a voip phone?? I'm trying to decide which on to buy. The
sl-6000 is perfect for phone use from what I've read, but it's not very
pocket friendly (http://www.sharpusa.com/images/hpc_SL6000_pic1.jpg)
The clamshell
most helpful thanks! :)
On Sat, 19 Mar 2005 16:01:31 -0500, Tyler [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
should tell u everything u need.
tf.
On Sat, 2005-03-19 at 15:51, Matt wrote:
Hi,
If I have a PRI card in my asterisk server
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