Hi, This app looks perfect for what I need. Are there any instructions how
to install?
- Original Message -
From: Dan Austin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 12, 2005 1:15 PM
Subject:
I just did this for a customer. All I did was create a queue just for him,
he is the only agent in the queue. * acts just like you want if you are the
only person in the queue.
- Original Message -
From: Jan Marius Evang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
I used agentcallbacklogin app with a extension macro, that way if he was on
a conference call he could log out and the caller would be sent to his
voicemail. Here is another approach though, it is in the WIKI and does
work
http://www.voip-info.org/wiki-Asterisk+tips+campon
- Original
1 Transcoding is between codecs. ulaw to
g.729 for example
2 I prefer AMP but unless you install it with
[EMAIL PROTECTED] it could be a pain.
3. You need a clock source for meetme and
other features to work so if you don't have any digium hardware you must use
ztdummy
4. Unless you are
What settings are you using to burn the iso? If you are using Nero or
several others you have to tell it to burn the disk at once, not track at
once. I had the same problem
- Original Message -
From: John Novack [EMAIL PROTECTED]
To: Scheda [EMAIL PROTECTED]; Asterisk Users
check for interrupt conflicts, cat /proc/interrupts
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, March 15, 2005 8:26 PM
Subject: [Asterisk-Users] Voice getting
Dan, Thanks for the time helping me out. I figured everything out except
for the patch.
7. cd to asterisk/apps and run patch -p0
path-to/apps-meetme-cbmysql.txt
When I do this step it errors out and asks for the file to patch.. When I
look at the apps-meetme-cbmysql.txt It shows the file
I installed this and it seems to be working great. Good job. Just one
question though, What is the shared extensions file?
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Steve can you post your Cisco configs? Can you post the configs from your *
box that pertain to your issue?
- Original Message -
From: Steve Blair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 16,
Check the WIKI there is an example of how to do this very thing.
I implemented it on a customer a few months back and it works great.
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config
- Original Message -
From: Luki [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
I replied with the right answer but for the wrong post, Sorry about that.
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 16, 2005 6:35 PM
Subject: Re: [Asterisk
My apologies to the list and to everyone that had bounced
messages sent to them from my email. Mediacom had an issue with my email
account it was bouncing messages saying it was full when it was not full. I apologize
again if this inconvenienced anyone.
Henry
My apologies to the list and to everyone that had bounced
messages sent to them from my email. Mediacom had an issue with my email
account it was bouncing messages saying it was full when it was not full.
I apologize again if this inconvenienced anyone.
Henry
forapp_cbmysqlandMeetMe2gui (out of tree modules)
Bugger. I knew I'd screw up the patch instructions.
Try this-
# cd to /var/build_aah/asterisk_src/asterisk
# patch -p1 path-to/apps-meetme-cbmysql.txt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent
That phone number is a Hendersonville, NC phone number.
According to the phone company records that is not the name of the person
the phone number is associated with.
- Original Message -
From: Vincent [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
It doesn't matter if I run it from the apps directory or the asterisk
directory I get the same response. This is getting frustrating.
- Original Message -
From: Dan Austin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
directories. Another option is to just edit
the apps-meetme-cbmysql.txt and split it into
three patchs and apply them one at a time.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Thursday, March 17, 2005 10:53 PM
To: Asterisk Users Mailing
Blibs are IRQ conflicts. cat /proc/interrupts
see if the IRQ of the X100P (wcfxo) is being shared with anything. If it
is try changing PCI slots
- Original Message -
From:
Reuben Grech
To: asterisk-users@lists.digium.com
Sent: Friday, March 18, 2005 9:59
AM
Interrupt 5 is the problem. Try changing slots on your x100p
- Original Message -
From: Reuben Grech [EMAIL PROTECTED]
To: 'Jason Williams' [EMAIL PROTECTED]; 'Asterisk Users Mailing
List - Non-Commercial Discussion' asterisk-users@lists.digium.com
Sent: Friday, March 18, 2005 11:54 AM
PROTECTED] On Behalf Of Henry
Devito
Sent: Thursday, March 17, 2005 10:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules)
It doesn't matter if I run it from the apps directory
TE110P works here for us. Here is the pinout
Pin 1-Receive (+) Ring 2
Pin 2-Receive (+) Tip2
Pin 4-Transmit (-) Tip1
Pin 5-Transmit (-) Ring1
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi, You said you are hooked to an analog PBX. Do you have to dial a 9 to
gain access to an outside line on the PBX. If you do the easiest thing to
do is go to the outbound-local context which is
[outbound-local]
exten = _NXX,1,Macro(dialout-default,${EXTEN})
exten =
yes, this also known as an E-1 PRI
- Original Message -
From:
Rob Nicholson
To: asterisk-users@lists.digium.com
Sent: Sunday, March 20, 2005 6:20
AM
Subject: [Asterisk-Users] ISDN-30 in
UK
Hi,
Ive just been made aware of
Asterisk through the
1. create a directory inside /var/lib/asterisk or whatever you have
configured for that, i.e. /var/lib/asterisk/mohmp3-radio, then
2. create /var/lib/asterisk/mohmp3-radio/dummy.mp3
3. then add
live =mp3:/var/lib/asterisk/mohmp3-radio,http://www.yourfavradio.com:port/
into your
Hi Ken, This has worked fine for me for about 6 months, maybe I just
didn't notice a problem. As far as I know there has been music playing when
people are being put on hold every time.
- Original Message -
From: Ken Godee [EMAIL PROTECTED]
To: Matt [EMAIL PROTECTED]; Asterisk Users
There was a bounty a while back to set up SMDI on *.This would be ideal
if you had a serial interface on your PBX. By the #63 and #64 code it looks
like you are talking about a Toshiba PBX. At one time I actually wrote a
cron script that would check to see if there were messages in
This will turn the lights on but not off~
- Original Message -
From: Brian S. Adelson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: David Brodbeck [EMAIL PROTECTED]
Sent: Tuesday, March 22, 2005 3:17 PM
Subject: Re:
Couldn't you do call files and have them drop the call into the meetme room?
- Original Message -
From: Alex Pepper [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 12:15 PM
Subject: [Asterisk-Users] Multicall
This is my question:
it is possible to
looks like you are missing the trailing '/' in the
url
- Original Message -
From:
Dov Bigio
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 12:38
PM
Subject: [Asterisk-Users] slim server for
moh
Hello,
I have installed
I'm doing the same scenario for 2 customers right now. Works good, On the
opt11 I connected it to asterisk with PRI. Have a good day
Henry
- Original Message -
From: Friend, George E. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 23, 2005 8:55 PM
Faxes I found are mostly unreliable if the padding is not set correctly on
the analog trunks.
The example below shows how to set the mailbox per timezone. Also what is
your system timezone set for? the tz option sets the zone.
;4200 = 9855,Mark
Spencer,[EMAIL PROTECTED],[EMAIL
: [Asterisk-Users] 2 [EMAIL PROTECTED] issues away from bliss
I have an x100p card is this padding something that can be set? I'm off
to google to see if I can find some fax padding. :)
Henry Devito wrote:
Faxes I found are mostly unreliable if the padding is not set correctly
on the analog
What do you get for an output from the CLI? Is the 9 being stripped?
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 24, 2005 9:16 AM
Subject: [Asterisk-Users] Question on
Search wiki for ASTTAPI
- Original Message -
From: Remco Barende [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Thursday, March 24, 2005 1:28 PM
Subject: [Asterisk-Users] Outlook contacts - Asterisk
database(LookupCIDName)
Is it possible in any way to
RIGHT FROM THE WIKI:
For simple dialplans first edit features.conf as desired, then put this into
your extensions.conf:
include = parkedcalls
If you have a more complex dialplan and want to be able to Goto() a more
elaborate 'parkedcalls' handler then you'll need to be sure to include a
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Jueves, 24 de Marzo de 2005 02:56 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outlook contacts -
Asteriskdatabase(LookupCIDName)
Search wiki for ASTTAPI
- Original Message
: Thursday, March 24, 2005 8:44 PM
Subject: RE: [Asterisk-Users] Outlook
contacts-Asteriskdatabase(LookupCIDName)
Which one? Didn't see it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Jueves, 24 de Marzo de 2005 04:39 p.m.
To: Asterisk
Search Google. This is not a key system it is
a pbx. I don't think you can accomplish what you want with
this.
- Original Message -
From:
Mark W Wood
To: asterisk-users@lists.digium.com
Sent: Friday, March 25, 2005 10:00
AM
Subject: [Asterisk-Users] Square Key
This can be accomplished if the last 3 of the number you want to send to the
outside match the extension by using variables.
- Original Message -
From: Sean A. Newton [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
I would you an ATA and something like a Viking door
box. Then if they ring the door bell it can call your phone and you could
speak to the person to tell them you are on your way, leave the package or
whatever.
- Original Message -
From:
Angus
Comber
To:
configuration (every C.O. line appears on every
phone)
Can I program a specific
C.O. line directly to a button?
Date: Fri, 25 Mar 2005
11:06:04 -0600
From: "Henry Devito" [EMAIL PROTECTED]
Subject: Re:
[Asterisk-Users] Square Key system
To: "Asterisk Users
Open loop Disconnect. AKA kewlstart!
- Original Message -
From: David Hill [EMAIL PROTECTED]
To: asterisk-dev@lists.digium.com; asterisk-users@lists.digium.com
Sent: Friday, March 25, 2005 6:33 PM
Subject: [Asterisk-Users] Openloop disconnect?
Hello there,
I tried to found
I've been working on, actually just started, creating a network app where
windoze pc's can print to a virtual printer which in turn will make asterisk
send the fax out.
I also have asterisk set up for a client where all it does is send and
recieve faxes. They have 14 fax machines on SPA2000
Here is the tab delimited you can import this into excel than export it as
coma.
http://www.areacode-info.com/COC/codedownload.htm
- Original Message -
From: Mark Halverson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Actually they have excel formatted files already and access database on that
site.
Henry
- Original Message -
From: Mark Halverson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Sunday, March 27, 2005 2:14 AM
Hate to bring up an old thread. I just configured a 7960 with multiple
lines appearing. Each defined the same, but the buttons don't seem to roll
over. What else do I have to define to do this.
Henry
- Original Message -
From: Chris Wade [EMAIL PROTECTED]
To: C F [EMAIL PROTECTED];
If you call Cisco contract support. 1-800-447-9347 and give them the serial
number used when you purchased the smartnet they will give you the contract
number over the phone. If the contract was sold properly the reseller would
have asked you for the serial number of the unit and turned that
] Cisco 7960 SIP images
Henry Devito wrote:
If you call Cisco contract support. 1-800-447-9347 and give them the
serial number used when you purchased the smartnet they will give you the
contract number over the phone. If the contract was sold properly
No serial number was asked for.. I just
calls
On Mon, 28 Mar 2005 10:31:56 -0600, Henry Devito [EMAIL PROTECTED]
wrote:
Hate to bring up an old thread. I just configured a 7960 with multiple
lines appearing. Each defined the same, but the buttons don't seem to
roll
over. What else do I have to define to do this.
Well it works for me
defiantly, This is one feature I've been trying to implement.
- Original Message -
From: Dan Austin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 8:15 PM
Subject: RE: [Asterisk-Users] [EMAIL
Title: Message
If you use AgentCallBack * doesn't keep the
call up.
- Original Message -
From:
dovb
To: asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 8:50
PM
Subject: [Asterisk-Users] call center:
agents, queues, sip
Hi,
I am doing some
Which sound file is the one you hear when you call
voicemail and it says Comedian Mail? I can't find it in the sounds
directory
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
I compiled and installed cbmysql.From the
command line if I do a show applications should I see cbmysql in that
list? I guess what I am trying to see is if cbmysql is connected to my
mwqsql. IS there anyway. I was hoping to be able to do it from *
CLI.
That's a lot of users for just a couple PRI's are you planning on doing IP
trunking too? Just a thought.
I proposed a approx 1000 phone system with 4 * boxes and 1 SER box for load
balancing. 3 of the * boxes I had the phones registering on the 4th was
used just for trunking.
Henry
-
What I have done is made a few different
configuration file scriptsand when I install asterisk I run which ever
script Ithink will work for the customer.IE instead of doing a
make samples I do a make system1.config or make system2.config
Henry
- Original Message -
From:
An additional fault in 1.0.7 When you log into voicemail and select advanced
options there are none. On previous versions it would ask if you would like
to send a message, etc.
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Forget this post I had a typo in my voicemail.conf file sendvoicemail=yes
was spelled wrong.
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 31,
issues
Henry Devito [EMAIL PROTECTED] writes:
Forget this post I had a typo in my voicemail.conf file
sendvoicemail=yes was spelled wrong.
That fixes point 1) What about the others?
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
After calls come in, it works fine, however, I notice that even when
SIP/602 is on the phone, Asterisk will still ring her. I believe its due
to
the fact that the phone support call-waiting. Is there anyway that I can
disable this support only on queues and ring the next extension in this
case,
- Original Message -
From: Irakli Natsvlishvili [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 4:52 AM
Subject: [Asterisk-Users] Manipulation based on SIP extension
Hello there,
How do I
you can also do a netconfig from the prompt.
- Original Message -
From: W. Kevin Hunt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 8:55 PM
Subject: RE: [Asterisk-Users] AAH 0.6 - Change Network
The zaptel driver has the 'D' Tone defined in the 'tones.h' file I am
trying to figure out what I can do with asterisk so it will recognize that
and do a HangUp.
- Original Message -
From: Brian Leyton
To: 'asterisk-users@lists.digium.com'
Sent: Thursday, April 07, 2005 5:04 PM
I had the same problem at one site. We could not receive faxes with spandsp
reliably. Our solution that seems to have worked with no problems so far
was to use a SPA-2000 to a fax machine.
- Original Message -
From: Kevin Brennan [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Does anyone have station monitoring working on the Snom 360 softphone?
I have Snom 360 softphone ext 360 and I want to monitor Cisco 7940 ext 301.
How do I configure my extensions.conf? I've tried going by the wiki but it
just doesn't seem to work.
Never mind I had a dumb typo.
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 10, 2005 5:12 PM
Subject: [Asterisk-Users] snom360 hint priority
Does anyone have
exit and asterisk -r
- Original Message -
From:
Abraham
WEI
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 11, 2005 3:30
AM
Subject: [Asterisk-Users] Can I exit from
asterisk console without stoppingasterisk?
If the answer is
Here's a complete system for $91 US. I use this box at co located office
with * and 10 SIP-841's and works great
http://www.hcditrading.com/Shop/Control/Product/fp/vpid/1377166/vpcsid/0/SFV/29664/rid/117517
- Original Message -
From: Ken Godee [EMAIL PROTECTED]
To: [EMAIL PROTECTED];
I have done this with CTX's you need analog cards in your ctx. and fxo
cards in the * servers. Email me off list I am a Toshiba Dealer.
- Original Message -
From: Stephen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 10:17 AM
Subject:
The Snom 220 and Side cars you can have up to 3 side cars on a 220 there are
20 buttons on each side car.
- Original Message -
From: Sean Kennedy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 1:42 PM
Subject: [Asterisk-Users] Line Presence:
Hi all
I am currently working on the coding to provide D tone disconnect. There is
a work around I am using right now at a few customer sites. I have done
this several times, interconnecting Toshiba to Toshiba PBX's and Toshiba to
other pbx's.
- Original Message -
From: Brian Leyton [EMAIL
, 2005 4:45 PM
Subject: RE: [Asterisk-Users] Toshiba CTX100 integration with PABX for two
site
Henry Devito wrote:
I have done this with CTX's you need analog cards in your
ctx. and fxo cards in the * servers. Email me off list I am
a Toshiba Dealer.
Good point - I missed the fact that he
4:45 PM
Subject: RE: [Asterisk-Users] Toshiba CTX100 integration with PABX for two
site
Henry Devito wrote:
I have done this with CTX's you need analog cards in your
ctx. and fxo cards in the * servers. Email me off list I am
a Toshiba Dealer.
Good point - I missed the fact that he doesn't have
In your zapata.conf set
usecallerid=nocallwaitingcallerid=no and immediate=yes. Remove the
Wait(0) and start your first priority with answer.
- Original Message -
From:
Scott
Wolfe
To: Asterisk-Users@lists.digium.com
Sent: Thursday, April 14, 2005 5:25
PM
: RE: [Asterisk-Users] Toshiba CTX100 integration with PABX for two
site
Henry Devito wrote:
You don't want to use RSTU2's unless you want echo. RSTU3's
are a little better but BSTU's are what you need.
Will I have the same echo problem with RSTU2 on a DK-424?
I don't think I have another choice
I found this to happen when a caller is leaving a message * lights the MWI
as soon as the message is being recorded. If the called person calls into
the * and listens to the message before it is done they here only a partial
message and the VM sends an empty attachment. Strange isn't it? I
I can help you. Email me off list. [EMAIL PROTECTED] or [EMAIL PROTECTED]
I am a Cisco Partner.
- Original Message -
From: Gary Guthary [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 15, 2005 6:18 AM
Subject: [Asterisk-Users] Slack 10 install - THANK YOU -
Email me off list I can help you.
- Original Message -
From: Dan Levine [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 15, 2005 8:52 PM
Subject: [Asterisk-Users] CISCO 7970
Does anyone know where to get
If you can't get in to the setting through the phone you have to go in
through the console port or set a tftp server up on the address that is
programmed in the phone and create a .cnf file with the password you want.
Email me off list and I can help you.
Henry
- Original Message -
This is an illegal copy of software. If Cisco securities track this sale
the seller and the purchaser could be fined several thousand dollars and
even face jail time.
- Original Message -
From: Bernardino Campos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Anyways this doesn't have the 7970 firmware on it.
- Original Message -
From: Bernardino Campos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 15, 2005 11:09 PM
Subject: Re: [Asterisk-Users] CISCO 7970
Search the wiki for campon. It may help point you in the right direction.
- Original Message -
From: Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 16, 2005 3:43 AM
Subject:
We usually put an Adtran CSU ACE inline with the asterisk box and the Voice
T1 that way if the LEC wants to loop a CSU they can. An Adtran CSU ACE just
passes the traffic as it receives it, there is no setup involved.
Henry Devito
Telephone Connection, Inc
Network Design / Implementation
Phone
Anybody know if there are any issues with these phones and * ? Does the MWI
work and such?
Henry Devito
Telephone Connection, Inc
Network Design / Implementation
Phone: 402.330.7510
Fax: 402.330.8586
Toshiba CTX/DK/Stratagy Certified
Cisco Certified Internetwork Expert (CCIE) Voice ( VoIP
Does anyone know Zultys ZIP 2 ip phone is fully compatible with asterisk?
Does the MWI and call transfer work correctly?
Henry Devito
Telephone Connection, Inc
Network Design / Implementation
Phone: 402.330.7510
Fax: 402.330.8586
Toshiba CTX/DK/Stratagy Certified
Cisco Certified
That's because to hook analog phones up to a port the port must be fxs. So for your situation you need a card with 2 FXO for CO lines and 2 FXS for regular phones.
---Original Message---
From: [EMAIL PROTECTED]
Date: 01/16/05 00:16:21
To: Asterisk-Users@lists.digium.com
Subject:
Sorry about the HTML post, I was sending from my laptop and forgot to turn
off html in outlook. Have a nice day.
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 16, 2005 12:50 AM
To: [EMAIL PROTECTED]
Subject: Re: asterisk-users list and html posts
18 on the BETA site has worked the best for me so far.
---Original Message---
From: Yair Hakak
Date: 01/18/05 02:55:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Best Grandstream firmware to use?
i've actually had reboot issues since
Good day,
I just downloaded the latest CVS and it will not compile. This is the error
I receive:
pbx_dundi.c:54:18: zlib.h: No such file or directory
pbx_dundi.c: In function `update_key':
pbx_dundi.c:1313: warning: implicit declaration of function `crc32'
pbx_dundi.c: In function
www.thirdlane.com has already written a close dsource webmin module. I have no idea how much it costs or how well it works.
---Original Message---
From: David Shaw
Date: 01/20/05 10:35:53
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Good evening. QSIG is the standard that ISDN PRI is written on Q.931. Q
SIGNALLING refers to the information provided on the D-channel. This
information creates ability to have transparent dialing plans and such. *
supports most of this functionailty already. I have this working on a
Very simple
exten = s,1,answer
exten=
s,2,Dial(SIP/${EXTEN1}(SIP/${EXTEN2}
- Original Message -
From:
[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 20, 2005 9:32
PM
Subject: [Asterisk-Users] Ring an
incoming call in multiple
Just download 1.0.3, don't use the voicepet scripts. they are versions 0.7
and 0.9 of *. If it is a true clone from digit networks it will work fine
with 1.0.3. I just set up 3 different machines with the cards from digit
networks.
Henry
- Original Message -
From: Dave Green
)
Henry Devito wrote:
www.thirdlane.com http://www.thirdlane.com has already written a
close dsource webmin module. I have no idea how much it costs or how
well it works.
I've attempted to contact thirdlane to get pricing on their GUI and
can't seem to get anyone to reply.
My
Hi, I have not implemented any of the spa-2000's yet. Do they work ok with
asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is
it two fxs ports with the same extension?
___
Asterisk-Users mailing list
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, January 22, 2005 4:54 AM
Subject: Re: [Asterisk-Users] Segmentation Fault after Digitnetwork
X100Pinstall
On Sat,
Is there ant chance of the Dialogic card model D/4PCI working with
asterisk ?
Word of caution: Even if you can buy the drivers and make this card work
with *, it is not meant to plug directly into a CO -48vdc talk battery and
90-130vac ring voltage delivered by your phone company. These
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, January 22, 2005 8:15 PM
Subject: Re: [Asterisk-Users] Dialogic D/4PCI
Henry Devito wrote:
Is there ant chance
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
- Original Message -
From: Erik Espinoza [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 23, 2005 6:40 PM
Subject: Re: [Asterisk-Users]
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
- Original Message -
From: Erik Espinoza [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 23, 2005 6:40 PM
Subject:
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
Do you have a contact email for these guys? I couldn't see anything listed
on their site anywhere. Seems the site is in current development.
Matt
Hi Matt,
I was just getting ready to try to order a IP
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