After having looked around on google, voip-info, etc, I am coming
here for a bit of insight from anyone who might be willing to provide
some
I am using * (it's [EMAIL PROTECTED], but this should apply to any * install) to
handle my home phone line. I have a TDM11B card, and my Verizon
I'm not sure if this is supposed to happen, but when I press the # key it
seems to have the effect of flashing the hook, or at least letting me
transfer. I am using Zap hardware.
Do you hear a transfer voice prompt? Asterisk will intercept the #
key if told to do so in the Dial application. In
Hi.
Probably been asked before, but my IAX provider assures me its not their problem
I have a IAX connection to a peer providing a DID. I am dialing up
my number, seeing the DTMF tones come down the line, and the * IVR is
just ignoring them.
IAX debug output is:
Rx-Frame Retry[ No] --
How come an outgoing call using my TDM400P immediately
say the call is answered? I'd like to be able to
detect when the call is actually picked up, is this
possible?
If this is normal with analog cards, does the same
thing happen with T1 cards?
-L
On Thu, Jun 30, 2005 at 06:56:16PM -0500, OMS wrote:
Hi,
I am running Asterisks on Public IP with Fedora Core 3.
What is the recommendation for making Linux secure on the Public IP since
I am new to Linux. Which Firewall should I use? I am not intending to
use Linux as router.
I wrote
On 6/30/05, Mark Charlton [EMAIL PROTECTED] wrote:
Hi
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
hi all,
i want to ask about caller id in asterisk.
if i make outgoing call, people who receive my call
can see the pilot number.
what i must do if i want that people receive my other
number (DID number)?
is it posibble?
thank to everybody..
regards,
shahdan
how can i solve the error on the last part?
need help. thnx...
Zaptel Configuration
==
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03:
hello all,
what i mean is, i make a call from another did number
but people receive the pilot number.
i don't know how to do :(
i try this but nothing happen.
exten = _01,1,SetCIDNum(0${CALLERIDNUM})
exten = _01,2,Dial(${TRUNK}/${EXTEN})
please help me..
regards,
shahdan
Hi all!
I am working with asterisk and trying to get it work in DHCP network, where
asterisk gets a DHCP address as well as other computers (IP-phones). So far,
I've have got asterisk working with static IP address where phones are getting
their IP from DHCP server.
Is it possible at all to
Hi,
-Original Message-
what i mean is, i make a call from another did number
but people receive the pilot number.
i don't know how to do :(
i try this but nothing happen.
exten = _01,1,SetCIDNum(0${CALLERIDNUM})
exten = _01,2,Dial(${TRUNK}/${EXTEN})
Hi,
I don't think it's goonna work that way !
The phones need to register with the asterisk server. So when you
configure the phones, you specify the ip of the asterisk server... so...
Another way, if you want to work with dynamic ip address would be to set
up a DNS (name resolver) which
Hi everybody,
I'm not a programmer, so I really don't know this,
but is it possible to somehow backport ${BLINDTRANSFER}
variable functionality to 1.0.X versions of *.
I need this really badly ... thanks for your replies.
Ivan
___
Asterisk-Users
You helped me, David, Thanks!
Jussi
Quoting David Masure [EMAIL PROTECTED]:
Hi,
I don't think it's goonna work that way !
The phones need to register with the asterisk server. So when you
configure the phones, you specify the ip of the asterisk server... so...
Another way, if you want
On Fri, 2005-07-01 at 10:15 +0300, [EMAIL PROTECTED] wrote:
Hi all!
I am working with asterisk and trying to get it work in DHCP network, where
asterisk gets a DHCP address as well as other computers (IP-phones). So far,
I've have got asterisk working with static IP address where phones are
Hi,
I have been
trying to enable attended transfer for callee. When
the callee pressed *2, DTMF tone was heard by the
caller. But when the caller pressed *2, attended transfer started. Its
strange.
I used two
SIP phones. My Asterisk version is Asterisk CVS-HEAD built by [EMAIL
Hi there!
I wan't to connect my * to an Alcatel 4400. Does anybody have some
experiences with that?
I have the problem that I can dial in to *, but not *-A4400 . :-(
Thx
Marc
___
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Asterisk-Users@lists.digium.com
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I try to
dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time out. It
seems it is not connected to the server.
Probably been asked before, but my IAX provider assures me its not their problem
I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them.
IAX debug output is:
Rx-Frame Retry[ No] -- OSeqno:
my extensions.conf
GROUPCALL = Zap/g2/12000221Zap/g2/12000222Zap/g2/12000223Zap/g2/12000224
exten = s,1,Dial(${GROUPCALL})
exten = s,2,Hangup
exten = s,102,Answer
...
The Problem is, the asterisk output and the Master.cvs tells me that the
Group was called but not who in the group answerd
Hello!
Does anybody know what will be the around price for the announced E3
card from Digium? When is it planned to be ready?
What kind of harware can be used for a system with such card which makes
only IVR stuff?
Thanks in advance!
Kind regards,
Tamas
On Fri, Jul 01, 2005 at 10:15:20AM +0300, [EMAIL PROTECTED] wrote:
Hi all!
I am working with asterisk and trying to get it work in DHCP network, where
asterisk gets a DHCP address as well as other computers (IP-phones). So far,
I've have got asterisk working with static IP address where
Hi,
I have been trying to configure my Asterisk to use a Sip provider for
out and incoming calls.
I only have one user and password for connect to my sip provider.
My sip.conf is:
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
Has anyone had any success with this card?
Thank you.
Sandy.
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To UNSUBSCRIBE or update options visit:
On Thu, Jun 30, 2005 at 10:13:18AM -0500, Moises Silva wrote:
Hi Bharat. I think that does not exists such a thing like an Asterisk
Dev App howto :p , so for now the best way to learn i think is check
out the apps/ directory on Asterisk Sources. Also check the app.h file
in includes/
in
Does the registration show up?
try sip show registry at the CLI
also try sip debug peer sip_proxy and post the result.
Might be able to see what's going on there...
mark
On 7/1/05, David [EMAIL PROTECTED] wrote:
Hi,I have been trying to configure my Asterisk to use a Sip provider forout and
BudgeTone Software
Version:
Program--1.0.5.11
Bootloader--1.0.0.18 HTML--1.0.0.37
VOC--1.0.0.6
I set the dtmf to
info and set via sip info on budgetone conf.page ,this time can not dial to
internal extensions of telephony systems of the company
Hello,
I want to recieve the output from astmanproxy in a php script.
Is that possible ?
I made a simple php script:
PRE
?php
$socket = fsockopen(127.0.0.1,1234, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: xxx\r\n);
fputs($socket, Secret:
Robert Goodyear wrote:
I'm sure you really only want to know about the absence of problems.
From watching this list for 6 months it seems the SuperMicro products
are most lauded and have exhibited no hardware conflicts. Various votes
on Dell products, so you're probably best to stay away,
Me wrote:
How come an outgoing call using my TDM400P immediately
say the call is answered? I'd like to be able to
detect when the call is actually picked up, is this
possible?
If this is normal with analog cards,
Yes
does the same thing happen with T1 cards?
No (unless you configure it
On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:
Has anyone had any success with this card?
Thank you.
There are a number of cards based on that chipset. I've had success with
one here, FWIW.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
setup. I have no special HW and intend to use asterisk on an internal
network just
Betül Gözlükoğlu wrote:
*I set the dtmf to info and set via sip info on budgetone conf.page
,this time can not dial to internal extensions of telephony systems of
the company*
Install version 1.0.6.6, you can get it at: http://gs-firmware.gratissip.dk/
Make sure you are set to:
/Send DTMF:
Tzafrir Cohen wrote:
On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:
Has anyone had any success with this card?
Thank you.
There are a number of cards based on that chipset. I've had success with
one here, FWIW.
Thanks, I assume these 4 cards dont work then. The
Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: 01 July 2005 01:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] make error for zaptel
Hi All
We are unable to hear any voice where as in tcpdum it
shows that RTP is flowing both ways
ERROR CONDITION- Executing
Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack --
Called 2000 -- SIP/2000-0ead is
ringing -- SIP/2000-0ead answered
SIP/2001-f6c4 -- Attempting
Ritesh Jalan wrote:
Hi All
We are unable to hear any voice where as in tcpdum it shows that RTP is flowing
both ways
ERROR CONDITION
---
-- Executing Dial(SIP/2001-f6c4, SIP/2000|20) in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered
Hi all,
Currently when someone leaves a voicemail the message is
stored in the /var/spool/asterisk/voicemail/default/(users ext)/INBOX ,
as it should. However Ive noticed that a copy is placed in the
/tmp directory. Once a message is heard and deleted, the copy in /tmp
remains. My
Do you know where to get one of these?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
chawki hammoud
Sent: Thursday, June 30, 2005 4:35 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] wi-fi phone advice
Hi:
I want to
Yes, I have :-)
3 of this cards running well on my personnal *
What price for your ?
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sandy Thomson
Envoyé : vendredi 1 juillet 2005 12:37
À :
On Friday 01 Jul 2005 13:08, Terry Wade wrote:
Ln -s /lib/modules/'uname -r'/build /usr/src/linux-2.6
Nope, I doubt that. The end user should read /usr/src/linux/README.suse
and see how to prepare the kernel for building thirparty modules.
-Original Message-
From: [EMAIL PROTECTED]
Tamas J wrote:
Does anybody know what will be the around price for the announced E3
card from Digium? When is it planned to be ready?
Pricing and release date have not been announced at this time.
___
Asterisk-Users mailing list
I need a Digium or Sangoma T1 card that has at least 2 spans on it
fairly quickly. Does anyone know of a vendor for either of these in NH
or Northern MA?
Please let me know!
Tom
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Try two different entries:
sip.conf:
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
callerid=No CallID
register =
http://www.voipsupply.com
or call at
1-800-398-VOIP
they can rush deliver if you need it.
Original Message
Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
near NH??
From: Tom Rymes [EMAIL PROTECTED]
Date: Fri, July 01, 2005 8:31 am
To: Asterisk Users
Sandy Thomson wrote:
Tzafrir Cohen wrote:
On Fri, Jul 01, 2005 at 11:36:31AM +0100, Sandy Thomson wrote:
Has anyone had any success with this card?
Thank you.
There are a number of cards based on that chipset. I've had success with
one here, FWIW.
Thanks, I
There's also http://www.atacomm.com.
On 7/1/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
http://www.voipsupply.com
or call at
1-800-398-VOIP
they can rush deliver if you need it.
Original Message
Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
Hello,
Either Digium or Sangoma can overnight a card to you. As the car drives you
could go to Toronto and pickup a card from Sangoma if you needed if a few
hours before Overnight would deliver it.
There are also a lot of resellers that can overnight to you as well.
MATT---
-Original
On Fri, 2005-07-01 at 08:01 +0200, Wilson Pickett wrote:
I'm not sure if this is supposed to happen, but when I press the # key it
seems to have the effect of flashing the hook, or at least letting me
transfer. I am using Zap hardware.
Do you hear a transfer voice prompt? Asterisk will
Hello everybody,
I have made a application of my
own. (I.e. Def ( )). I am able to compile the application successfully. And the
.so file is created as well. But when I load asterisk I get the following
error.
[Def.so]Jul 1 19:20:06 WARNING[15664]: loader.c:295
ast_load_resource: No
I do not want to use the default key of '#' for call transfer, because
as we all know, it interferes with many IVRs that require # as a
termination character. I modified features.conf and added:
[featuremap]
atxfer = **
The double-star now works great. If I press it while on a call, I go
into
hello
i am trying to follow
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
can any one tell how to install this
2. Install Asterisk::AGI and Asterisk::Manager
(unfortunately it is not on CPAN yet!)
thanks in advance
Kamran
Hi men,
If you unplug a telephone line behind X100P/X101P, you must have an alarm
about this Zaptel device on the Asterisk console.
When you plug the line, you can see alarm stop on a new line (I don't
remember the exact message).
Very nice test.
Best Regards,
Francois BERGERET,
France.
Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first line of my message) I
specifically rebooted the box. Have to confess I am really flumuxed why
the symbolinc link differs
Hello,
I am
using asterisk to autodial a phone number once an hour to verify an answering
IVR scriptis working by listening for a beep played by the IVR, and that
is working well. The one thing I have not been able to find is a way to
trap if the call didn't connect at all, like if the
This unit is vaporware from what I can tell.
Cory Andrews
Purchasing / EVP
VOIPSupply.com
v – 716.630.1555 X22
e – [EMAIL PROTECTED]
Huddleston, Robert wrote:
Do you know where to get one of these?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Dear list members,
I am trying to use mysql for the mailbox definitions of the voicemail
system. All works fine, but my asterisk does not catch the options of
the mailbox. In particular, 'attach=no' has no effect. Here is my test
table:
mysql select * from users where email='[EMAIL PROTECTED]';
Sandy Thomson wrote:
Has anyone had any success with this card?
Thank you.
I am looking for a source for the clones in NZ - getting the real deal here isnt
an option (killer shipping) and at the moment I am just having a play with
asterisk and have given up on the internet linejacks I
I know that CW can be turned on and off with database put CW which updates
my /var/lib/asterisk/astdb.
However, I would also like to be able to configure this through the flat
config files in my /etc/asterisk, is this possible? I'm using
CVS-v1-0-06/24/05-14:34:05.
TIA for any help.
--
Tim
Dinesh,
This should be fine as long as you set canreinvite=no for both systems
in sip.conf on Asterisk. I've done both Asterisk - SER and Asterisk
- CCM, and they work well.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Dinesh wrote:
Hi all,
I have
[EMAIL PROTECTED] wrote:
Hi men,
If you unplug a telephone line behind X100P/X101P, you must have an alarm
about this Zaptel device on the Asterisk console.
When you plug the line, you can see alarm stop on a new line (I don't
remember the exact message).
Very nice test.
Best Regards,
Francois
My one from www.broad-tel.com works fine and is very cheap.
On 7/1/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Yes, I have :-)
3 of this cards running well on my personnal *
What price for your ?
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL
Well poo - if I can use that word I'm one of those poor family guys who
loves to buy hardware on the cheap =)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Cory Andrews
Sent: Friday, July 01, 2005 10:26 AM
To: Asterisk Users Mailing List -
Robert - I suspect what they are doing is just trying to build buzz
and simply not mentioning the fine print here. There is no way, given
manufacturing and importation costs, that I can believe they can build a
solvient business offering around a $39 WLAN phone selling just the
hardware, that
If it does materialize, im up for 3 or 4 of them at that price.
Huddleston, Robert wrote:
Well poo - if I can use that word I'm one of those poor family guys who
loves to buy hardware on the cheap =)
smime.p7s
Description: S/MIME Cryptographic Signature
On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first line of my message) I
specifically rebooted the box. Have to confess I
Mike Myers wrote:
Hi.. I am about to replace my aging Nortel Venture
system with an Asterisk system and 6 Polycom IP 501
phones, and a couple sipura 841's for less used areas.
We have 3 phone lines here. One is SBC, one Vonage,
and one Voipjet... One hangup is that I can't figure
out how to
Hi list, i'm an asterisk newbie and i've to setup a net with an
asterisk server and several ip phones linked on the net.
i hope my questions are IT ans if you have some link for solving those
problems please mail me.
i've wrote the sip.conf in this way:
[2011]
type=friend
username=2011
secret=1945
Bob Goddard wrote:
On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first line of my message) I
specifically rebooted the
We have some guys working in the US who can't always dial back to our
company in Europe easily (lots of clients require authorization to make
international calls), so I set up the following:
- ipkall.com number links to a FWD number
- office Asterisk box registers with FWD
Then I
may be reading include/asterisk/module.h you can find the answer. May be this?
/*! Returns the ASTERISK_GPL_KEY */
/*!
* This returns the ASTERISK_GPL_KEY, signifiying that you agree to the terms of
* the GPL stated in the ASTERISK_GPL_KEY. Your module will not load if it does
* not return
Darren Wiebe wrote:
Could you please post the output from the asterisk console when
astcc.agi crashes? I really would like to get this resolved.
Darren Wiebe
[EMAIL PROTECTED]
Darren here I post you the output from asterisk console and the mysql
daemon log. After hanging the phone the field
I have put a pbx into a resort with Polyycom phones, everythign works
great, except the kitchen staff cannot hear the phone ring. I know many
legacy systems employ a big red flashing light, any ideas on doing
something similar?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
On 7/1/05, Wilson Pickett [EMAIL PROTECTED] wrote:
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
I
How good is your electrical engineering?
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent:
You should be able to do a good job with IPTABLES which is included in
FC3. You can limit source destp IP and protocol, etc.
Type man iptables | more for more details...
OCG
-Original Message-
From: Terry H. Gilsenan [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 30, 2005 8:11 PM
John Novack wrote:
Mike Myers wrote:
Hi.. I am about to replace my aging Nortel Venture
system with an Asterisk system and 6 Polycom IP 501
phones, and a couple sipura 841's for less used
areas.
We have 3 phone lines here. One is SBC, one Vonage,
and one Voipjet... One hangup is that I can't
On Wednesday 29 June 2005 23:36, Michael Stahl wrote:
How have you found the quality (Choppy / smooth audio)?
Any problems registering? (I have been unable to register for hours)
I use them for some of my termination, they seem to work just fine (no
quality/registration issues).
Actually
I have put a pbx into a resort with Polyycom phones, everythign works
great, except the kitchen staff cannot hear the phone ring. I know many
legacy systems employ a big red flashing light, any ideas on doing
something similar?
FYI, the Uniden UIP200 has a big red flashing light.
Analog relay in the same ring group with a bell?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
Sent: Friday, 1 July 2005 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I try
to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote:
Robert Goodyear wrote:
I'm sure you really only want to know about the absence of problems.
From watching this list for 6 months it seems the SuperMicro products
are most lauded and have exhibited no hardware conflicts. Various
I haven't heard much feedback yet - anyone here using VocTel?
The connection problem turned out to be my firewall, but I'm curious if
others experience any voice choppiness or high latency. Some posters
have related the problem to specific VOIP providers, some seem to be ISP
related (local
Anyone know a good distro for an Epia Mobo with the C3
chip?
I have been trying to get Debian and Gentoo installed (new
to me) and so far having little luck.
Does anyone know a good install for this processor/mobo
combo?
Thanks
Wiley
Mike Myers wrote:
John Novack wrote:
Mike Myers wrote:
Hi.. I am about to replace my aging Nortel Venture
system with an Asterisk system and 6 Polycom IP 501
phones, and a couple sipura 841's for less used
areas.
We have 3 phone lines here. One is SBC, one Vonage,
Wiley Siler wrote:
Anyone know a good distro for an Epia Mobo with the C3 chip?
I have been trying to get Debian and Gentoo installed (new to me) and
so far having little luck.
Does anyone know a good install for this processor/mobo combo?
You have to compile without mmx and
Finally, We have lift off, a shaky one though.
I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST.
I say "something" because the amount I expected and amount entered are different.
Please see below, I
I had the same problem in a welding lab. I did this on a 3Com NBX, but
I'm sure that the same idea would apply to Asterisk: I went to
RadioShack, and bought one of their visual ringers, for the hearing
impaired (basically flashes a white strobe light, and sounds a really
loud ringer), and
Oliver,
Thanks for the response! Do you know where I can find an example of how
to do this? I have never had to install a custom kernel before.
Thanks!
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oliver
Rath
Sent: Friday, July 01, 2005
Wiley Siler wrote:
Oliver,
Thanks for the response! Do you know where I can find an example of how
to do this? I have never had to install a custom kernel before.
For Gentoo there is a superb dokumentation on
http://www.gentoo.org/doc/en/index.xml to do this.
Regards,
Oliver
hello u can see the readme.udev in the zaptel directory that's normally answers
ur question
From: [EMAIL PROTECTED] on behalf of Ian Bert Tusil
Sent: Fri 7/1/2005 9:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Got this error after my
On 01/07/05, Mark Charlton [EMAIL PROTECTED] wrote:
I have been fighting with the Bayham Systems FastSMS AGI script, and I
re-wrote it as a stand alone Perl script. I am now calling it with
the EXTERNNOTIFY option in the voicemail.conf file. It gets passed
the context, extension and number
I have searched quite a few places and have not seen this discussed.
Basically I was wondering how would you go about having an option for
a user to be notified every 15 minutes until their new voicemail
message is checked. Since the notification e-mails we send get sent
to cell phones or actual
Ade Agbero wrote:
Finally, We have lift off, a shaky one though.
I deleted my Astcc.gi and replaced it with Darren's copy posted on his
website and I have finally been able to get something recorded as BILLCOST.
I got it working too here with Darren's astcc.agi. And billing as
expected
i tried to write to usa destination 1* it worked well
but when i tried to specify the number of digits i
wrote
1NXXNXX but it did'nt work.can anybody help me
please
please.
Yahoo! Sports
Rekindle the Rivalries. Sign
Hi,
We've exhausted our internal capabilities as well as Sangoma tech support and
were hoping someone with some expertise could help us with a pointer. Briefly,
our issue is as follows.
Periodically (several times an hour), we get either of the following error
messages in our asterisk messages
Pattern-matching extensions must be prefaced with an underscore thus:
_1NXXNXX
Enjoy!
On Fri, 1 Jul 2005 10:32:46 -0700 (PDT), wassim darwish
[EMAIL PROTECTED] said:
i tried to write to usa destination 1* it worked well
but when i tried to specify the number of digits i
wrote
You could have just done ln -s asterisk-1.0.9 asterisk and it would
have fixed that. It should by default do -I../asterisk
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote:
maybe zaptel verion incompatability try other newer or stable older versions
not sure thats just a hint
From: [EMAIL PROTECTED] on behalf of Zoltan Szecsei
Sent: Fri 7/1/2005 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
hi do u have the sip phones extensions in the extension.conf and are they in
the right context (sip-incoming)???
are the sip phone registering to asterisk?? try stop asterisk and reconnect as
asterisk -vvvc to check see them registering...
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