I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using
CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I
try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
I just
I had GXP-2000's running on 1.0 versions of asterisk even earlier. So I
know it does work. I upgraded one of my customers GXP's to the latest
firmware in it still works. Can you post the output of the CLI with verbose
set to 99 and the the output from the asterisk log file that has the call i
Is your phone actually registered to the switch. go to the CLI and do a
'sip show peers' see if extension 502 is registered. Making an outbound
call does not prove that the phone is registered.
- Original Message -
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - N
Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is)
I've had to add the voicemail context to get MWI to work correctly in the
past.
- Original Message -
From: "Jaap Winius" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, February 13, 2008 12:45 PM
Subject: [asterisk-users]
I am a Toshiba dealer and can help you do what you want to do. Do you
have qsig licenses in the Toshiba's now? If you do why not just use
toshiba's IP cards? anyways if you just want to integrate * to the
toshiba an want to see caller ID you have to use PRI cards not T1
cards. Email me directly
There is a version 2 firmware coming out the first week of
next month that you can upgrade the phone to. It is supposed to address a
lot of issues
- Original Message -
From:
Christopher Kenna
To: asterisk-users@lists.digium.com
Sent: Sunday, May 08, 2005 7:38 AM
Su
was: Re:
[Asterisk-Users]7960'multi-line' configuration
Henry Devito wrote:
---
I've since migrated away from [EMAIL PROTECTED], but the codes are there in the
extensions.conf file.
I am also cusrious to know as to what you have migrated to, from
[EMAIL PROTECTED]
ing I should
be
messing around with or should I be looking elsewhere?
Thanks!
Pat
Quoting Henry Devito <[EMAIL PROTECTED]>:
It has something to do with the AGI script. Scroll down!
- Original Message -
From: "Patrick M. Gray, Jr." <[EMAIL PROTECTED]>
To: "Asterisk
Are you using asterisk @ home?
- Original Message -
From: "Patrick M. Gray, Jr." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, May 03, 2005 9:22 PM
Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration
I still can't get the "mul
It has something to do with the AGI script. Scroll down!
- Original Message -
From: "Patrick M. Gray, Jr." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, May 03, 2005 9:22 PM
Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration
Nortel and Toshiba and so on help eliminate this by routing outgoing calls
starting from the highest trunk backwards and incoming calls of course start
from the lowest trunk and work upward.
- Original Message -
From: "Ryan Courtnage" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List
Usernames the same. The Cisco phone recognizes the usernames are the same
and only registers once.
- Original Message -
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users"
Sent: Monday, May 02, 2005 8:36 PM
Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration
You
It is only registering with asterisk once. ASterisk doesn't know that it is
a second button on the phone, the sip software on the phone makes the call
roll to the next free button.
- Original Message -
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Comm
- Original Message -
From: "Scott Henderson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, May 02, 2005 4:01 PM
Subject: Re: [Asterisk-Users] 7960 "multi-line" configuration
You can't use the same extension on multiple line buttons but
when
the call ends.
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Monday, May 02, 2005 12:39 PM
Subject: Re: [Asterisk-Users] Asterisk as VM for Nortel System
On May 2, 2005 01:01 pm, Henry Devito wrote:
What type of Nortel system? Is i
What type of Nortel system? Is it an option or a norstar?
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Monday, May 02, 2005 10:16 AM
Subject: Re: [Asterisk-Users] Asterisk as VM for Nortel System
On May 2, 2005 11:07 am, Matt wrote:
Can anyone think of a
From the CLI if you do a iax2 show registry, does it show you registered?
Maybe you can post the parts of your config that pertains to your question?
- Original Message -
From: "Patrick Gray, Jr." <[EMAIL PROTECTED]>
To:
Sent: Friday, April 29, 2005 11:03 PM
Subject: [Asterisk-Users] Can
You mention the WIP-5000, Does that handset have the ability to receive
text messaging/instant messaging?
- Original Message -
From: "Michael Graves" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, April 29, 2005 4:14 PM
Subject: [Asteris
- Original Message -
From: "Jacob Cazzell" <[EMAIL PROTECTED]>
To:
Sent: Friday, April 29, 2005 4:21 PM
Subject: [Asterisk-Users] Paging and intercom
On our existing phone system, if you dial an
extention the other end will beep and then setup an intercom channel
that's hands free for t
- Original Message -
From: "Jan Johansson" <[EMAIL PROTECTED]>
I seem to get "bounces" on DTMF.
For instance, if I turn on debug, and I dial the voicemail, and >enter 1234
as extension, I see in the logs "12234" "111234" "12344" and so >on, same
with passwords.
What type of phone SIP or
If you go to the fcc.gov website and search for CALEA there is around 7
documents that come up for April 27 2005. I believe I remember reading it
one of those documents.
- Original Message -
From: "Brian Capouch" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Disc
You ar making this a lot harder than it is.
If your incomming trunk is in the same context and there is an extension that
matches the DID the did will route to the correct place. Matter of fact I
always try to match DID to extension. Usually I have the telco only send
me the last 4 of the
line. N0
transfer,
- Original Message -
From: "Eric Wieling aka ManxPower" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 28, 2005 12:26 PM
Subject: Re: [Asterisk-Users] Call transfer
Henry Devito wro
I just bought one of these zyxel wireless phones, of course there is no
transfer key. Is there a patch for the stable 1.0.7 that I can implement #
or any other key or combination to initiate a transfer?
I looked briefly through the wiki and archived lists and didn't see much.
You need to change the line type in either your zapata.conf or your
zaptel.conf they need to match.
- Original Message -
From: "Luz Lopez" <[EMAIL PROTECTED]>
To:
Sent: Thursday, April 28, 2005 11:01 AM
Subject: [Asterisk-Users] start asterisk
Hi All.
I have installed Asterisk on linux
- Original Message -
From: "Isamar Maia" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, April 27, 2005 7:54 PM
Subject: Re: [Asterisk-Users] Linksys/Cisco buys Sipura
I guess the prices will go up like a rocket
Not necessarily, W
Look at the agents.conf. There is an option there to record calls. Maybe
this will point you in the right direction.
- Original Message -
From: "Daniel Salama" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, April 27, 2005 6:32 AM
Sub
Sounds like a possible disconnect issue, When the lines are not available
try doing a 'zap show channel X' with X being the channel number 1,2,or 3
and see if asterisk thinks the line is onhook or offhook. Just a thought.
Henry
- Original Message -
From: "Colin E. McDonald" <[EMAIL PR
Not that I know of I am a Cisco partner and the
Category 1 contract is still at least half that or less.
- Original Message -
From:
Dan Levine
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, April 27, 2005 8:49
AM
Subject: [Asterisk-Us
eaker echo question
Hmm..We currently have the 3com NBX system with VoIP which do echo back
in the earpiece. I thought it might just be the Cisco phone itself.
Hopefully soon I can test with some more Cisco phones to see if it is the
phone itself, or something else.
On Tue, 26 Apr 2005, Henry
This echo is known as side tone it happens naturally on analog lines, IP
phones usually do not provide this,
Henry
- Original Message -
From: "Jeremy Koski" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, April 26, 2005 1:53 PM
Subject:
You can only set 1 distinctive ring if by caller id. There is a tool on the
website to record custom ring tone.
- Original Message -
From: "Tomas Florian" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, April 26, 2005 1:39 AM
Subject:
AIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, April 24, 2005 11:09 AM
Subject: Re: [Asterisk-Users] TE11OP -> Mitel 200Sx??
Thanks Henry,
-Scott
- Original Message -
From: "Henry Devito" <[EMAIL PROTECTE
hat your telling me then I have to do D4/AMI. So does my zaptel
look correct? Maybe my cableing is off.
Thanks,
-Scott
----- Original Message -
From: "Henry Devito" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion&
Depends on the state you are in. In Nebraska there is no law saying you
have to tell someone they are being recorded if you are recording them on a
business line. In Iowa you don't have to tell them , but you have to play a
tone in the background every so many seconds.
- Original Message
When you do a sip show peers from the what IP address does it show for the
841?
- Original Message -
From: "Brian Watters" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Saturday, April 23, 2005 12:52 AM
Subject: [Asterisk-Users] IP Phones and
It can use DNS if the DNS servers are valid. Can you post your SIP.conf?
Didi you configure the phone manually or did you use the cnf files? If you
used cnf files can you post those also?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.c
Do you have nat=yes in the sip.conf? or alternatively do you have the phone
set for nat with the correct external ip address?
- Original Message -
From: "List Receiver" <[EMAIL PROTECTED]>
To:
Sent: Friday, April 22, 2005 10:49 PM
Subject: [Asterisk-Users] Cisco 7960 won't register as S
*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to
dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 200
You can get around this by creating the SIPDefault with only the first line:
image_version: "P0S3-07-3-00"
and a blank SIP
Email me off list if you need help
- Original Message -
From: "Gregory Wiktor - ADCom Corp." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing Lis
Looks like your signaling is wrong for I thought
PRI on Mitels had to be B8ZS, ESF. I could be wrong though. I haven't
worked on a 200 for a year or so.
- Original Message -
From:
Scott
Wolfe
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List -
Non-Commercial Discus
I was wrong. I just looked in my Mitel
I&M's. What level software are you on in the SX200? Up until a
certain level 200's could only do D4/AMI T1's, they could not do PRI's. If
it is a newer switch within the past 3 years or an older switch with later
software than you can do PRI, but th
Don't you need one of these directives so the PRI
knows which is master and which is slave?
pri_cpe: PRI signaling, CPE side
pri_net: PRI signaling, Network side
Henry
- Original Message -
From:
Scott
Wolfe
To: Asterisk-Users@lists.digium.com
Sent: Friday, Ap
That solution does exactly what I asked BUT! I'd like it to dial and
know which extension I'm coming from and then prompt for the password
> exten => _8501,2,VoicemailMain(s${CALLERIDNUM})
Just remove the 's' from the line above. Not to sound like a smart ass, but
this is all very well document
I have three in one machine, and 4 customers that have 2 in each of their
machines. The only problem I've ever had is momentary echo when a call
first begins, but that is to be expected until the line trains.
- Original Message -
From: "Paul" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mai
-- Executing Dial("SIP/3001-e13a", "ZAP/1/65869804") in new stack
This is what's wrong I think. The line is missing the 'g' for the trunk
group. On all of my [EMAIL PROTECTED] boxes the cli shows
-- Executing Dial("SIP/227-a4dd", "ZAP/g0/3428463") in new stack
___
Title: One touch voicemail on Cisco 7940/60
Yes it is possible. Just setup and extension
that logs into voicemailmain without asking for username or passcode. Then
set the messages configuration to that extension in the appropriate config file
on the 7960.
- Original Message -
on how to do this? Is this
done in queues.conf or in the dial plan? Even pseudocode will help.
Thanks,
Daniel
On Apr 21, 2005, at 2:55 PM, Henry Devito wrote:
3) When callers call into the * box and the agents are busy, they will
be put on the queue. Now, I wish to be able to tell the callers
Don't you have to configure your dialplan to hunt to the next extensions?
How else would * know to try 94207 if 4207 is busy?
- Original Message -
From: "kurt x" <[EMAIL PROTECTED]>
To: "Asterisk"
Sent: Thursday, April 21, 2005 3:08 PM
Subject: [Asterisk-Users] Multiple Line config help
- Original Message -
From: "Daniel Salama" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 21, 2005 9:31 AM
Subject: [Asterisk-Users] Queues configuration
1) If I understand correctly, an agent can belong to more than one queue.
Update further...The first conversation on the phone had the beeping as
described above, it occurred throughout the conversation. All of the
subsequent calls made (about 5, all less than 5 minutes) have been crystal
clear and I couldn't tell the difference from that or my POTS phone
connected to th
The whole system is running on an older dual processor PII 450Mhz machine
with SCSI drives.. 512Mb ram.The system runs RH9 with asterisk version
SCSI drives cause beeping too do to the demand for interrupts!!! With
IDE drives you can give the processes a low priority but those SCSI drives
th
You have to do a flash on the Siemens which gives you * dialtone then Dial
*0 which flashes the line. So the steps are flash *0
- Original Message -
From: "Sascha Ferley" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, April 20, 2005 6:32 PM
Subject: [Asterisk-Users] Call waiting
Hi,
I am tr
Can you post your config's? What version of * are you using? This doesn't
(Bhappen on any of my queues. I have queues set up on several customers
(Bsystems. If there are agents/members available the caller rings them
(Bdirectly, no announcements played.
(B- Original Message -
(
I am already doing this with AGI, PERL, and PHP to set up the page groups.
I will release the code as open source if people are interested. I'm not
the best PERL scripter in the world but it works.
- Original Message -
From: "Richard" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing Li
7:57 AM
Subject: RE: [Asterisk-Users] VPN/Asterisk combo
How do you get 18 interfaces on one machine?
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Henry Devito
Sent: Tuesday, April 19, 2005 8:26 AM
To: Asterisk Users Mailing
I use monowall in an executive building with 18 different LANS all ran
through the same firewall for Internet and IP phones with dual asterisk
servers.
- Original Message -
From: "Chris Mason (Lists)" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
S
You need a compatible sound card for the Dial and Hangup command to work.
- Original Message -
From: "Dan Levine" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 18, 2005 6:57 PM
Subject: RE: [Asterisk-Users] DIAL FROM CONSOLE
It say
Hi,
I am having a few issue withs [EMAIL PROTECTED] 0.9.
1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro)
softphone. I can receive voicemail no problem and even in this revision
the MWI seems to work correctly, though when i try to go to the message
center, (*98) and ente
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"
Sent: Saturday, April 16, 2005 3:43 AM
Subject: [Asterisk-Users] Extensio
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"
Sent: Friday, April 15, 2005 11:09 PM
Subject: Re: [Asterisk-Users] CISCO 7970
http://cgi.ebay.com/ws/eBay
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 Dis
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 -
Fro
Email me off list I can help you.
- Original Message -
From: "Dan Levine" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, April 15, 2005 8:52 PM
Subject: [Asterisk-Users] CISCO 7970
Does anyone know where to get the default firmware for th
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:
Sent: Friday, April 15, 2005 6:18 AM
Subject: [Asterisk-Users] Slack 10 install - THANK YOU - & Cisco
ResellerHelp
Hi Folk
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 wi
2005 6:38 PM
Subject: 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-
In your zapata.conf set
usecallerid=no callwaitingcallerid=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
, April 14, 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
;"
Sent: Thursday, April 14, 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
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" <[EMA
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:
Sent: Thursday, April 14, 2005 1:42 PM
Subject: [Asterisk-Users] Line Presence:
Hi all
With the recent thread on l
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:
Sent: Thursday, April 14, 2005 10:17 AM
Subject: [Asterisk-Users] Toshiba CTX100 integrat
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
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
Never mind I had a dumb typo.
- Original Message -
From: "Henry Devito" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, April 10, 2005 5:12 PM
Subject: [Asterisk-Users] snom360 & hint priority
Does anyone h
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.
_
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
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
Subject
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"
Sent: Monday, April 04, 2005 8:55 PM
Subject: RE: [Asterisk-Users] AAH 0.6 - Change Network Gateway
For this CentOS
- Original Message -
From: "Irakli Natsvlishvili" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 04, 2005 4:52 AM
Subject: [Asterisk-Users] Manipulation based on SIP extension
Hello there,
How do I configure any type of action base
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
I checked out asterisk sounds from the CVS and now all my audio files work
that I know of.
- Original Message -
From: "Bruno Hertz" <[EMAIL PROTECTED]>
To:
Sent: Thursday, March 31, 2005 12:42 PM
Subject: Re: [Asterisk-Users] Asterisk-1.0.7 Build - Serious issues
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"
Sent: Thursday, March 31, 2005 9:04 AM
Subject: [Asteri
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 -
What I have done is made a few different
configuration file scripts and when I install asterisk I run which ever
script I think 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:
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
- Ori
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.
___
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
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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
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"
Sent: Monday, March 28, 2005 8:15 PM
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released
I'm
alls
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.
Wel
ers] 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
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 i
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 PROTEC
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'"
Sent: Sunday, March 27, 2005 2:14 AM
Subject: [Asterisk-Users] NPA NX
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'"
Sent: Sunday, March 27, 200
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 t
Open loop Disconnect. AKA kewlstart!
- Original Message -
From: "David Hill" <[EMAIL PROTECTED]>
To: ;
Sent: Friday, March 25, 2005 6:33 PM
Subject: [Asterisk-Users] Openloop disconnect?
Hello there,
I tried to found documentation about openloop disconnect on
Asterisk/Zaptel. And u
e set 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: "
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