What the hell? NO!
show application MySql
app_addon_mysql is the name of the module.
load app_addon_mysql.so
-Matthew
Quoting Ronald Wiplinger [EMAIL PROTECTED]:
Matthew Boehm wrote:
Ronald_Wiplinger wrote:
I would like to put / get some data from an MySQL database.
I want to use this
You should take a look at Section 2.1.5 of
http://www.faqs.org/rfcs/rfc3435.html
This is the basis for the Polycom digit maps.
At 07:31 AM 7/28/2005, you wrote:
I am having a strange problem with polycom 501 and dailing. I've read the
archives and no one there specifically mentions this
Hi
I have configured sip accounts and they work some times. when i make a call
to another SIP account it works right
but some times i get the following error
Jul 29 07:17:00 WARNING[802]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Critical
Apologies for being a bit of a Linux
newbie...
I have got a working * system but each time I
reboot my box I need to:
modprobe qozap
ztcfg
asterisk
Now I realise this is really a Linux question but I
am struggling with the problem and any help would be much
appreciated.
There is a
Is there some configuration or specific extension in the asterisk to send
instant messages between two SIP clients? I'm using the eyebeam and this
service is not working!
Thank you.
[]'s
Wendell
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Brian West [EMAIL PROTECTED] wrote:
Digium, the creator and primary developer of Asterisk, the industrys
first Open Source PBX, will be hosting a pizza party from 4pm to 6pm
on the first day of Cluecon. We look forward to everyone coming out
to enjoy this opportunity to meet fellow developers
Christian Stredicke wrote:
It would be nice if the PBX can acknowlegdge the Record header - then it
would have the chance to paint a record icon on the screen.
In the next release.-)
Right.
Is there another header for turning off recording?
Anyway, we should not send unsupported media
Hi Jerry,
this is the cdr result of four tests (sorry for bad format):
Here is a crude hack, but it requires the user to press # at the end.
exten = s,1,Playtones(dial)
exten = s,2,Read(1stnumber,,1)
exten = s,3,StopPlaytones
exten = s,4,Read(restofnumber)
exten = s,5,SetVar(totalnumber=${1stnumber}${restofnumber})
Hope that helps.
B. J.
-Original
I've gotten CME to talk to *, but have not used the plain Call Manager.
I'd guess you could use a SIP trunk like the wiki talks about to configure
call managed to talk to a SIP termination service.
Rick
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Thursday,
I will be out of the office starting 07/29/2005 and will not return until
08/09/2005.
Thank you,
Marc
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Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything else but
no luck on the agent part..
___
Asterisk-Users
Greetings,
I have a Sipura SPA-1001. When I make outgoing calls, I have very
jittery sound. Incoming calls work fine. This wasn't the case a few
months ago, I am running head as of yesterday.
Any suggestions?
Thanks,
Erik
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I have not been receiving mail from the list 29th July, what is the problem
with gmail and the list.
No problem here.
Check you Spam folder, and if you find email there from this list,
select them all and click Not spam
hth
___
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Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
rhuddleston.vcf
Description: Binary data
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I can confirm that my IP 600 phone will reboot on the slightest
electrical glitch, even though it does not affect any servers,
workstations, other phones or any other equipment. I think the IP 600s
are very close to the maximum power output of the PSU. Perhaps the
easiest solution would be a
Haven't seen email since the 29th.. just testing.
begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWire Networks
adr;dom:;;4577 Pecos St;Denver;CO;80211
email;internet:[EMAIL PROTECTED]
title:Sr. Network Engineer
tel;work:303-458-5667 x 106
tel;fax:303-458-5725
x-mozilla-html:FALSE
OK, now this should be really simple, but I am a bit of a newbie so please bear
with me. I have an [EMAIL PROTECTED] box setup with TDM04B and two POTS lines.
On the
SIP side, I have GXP2000 phones. Most things seem to work, but the users
cannot figure out how to transfer incoming calls
I have a TDM400P with one FXS and one FXO..
how many liscence(2) I will have to buy?
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Hi Guys,
I thought some of you might be interested in a minimalistic Polycom
ringtones howto.
I assume this works with the ip600 (501/601) but not sure about the 300.
http://www.voipphreak.ca/archives/82-My-Little-Howto-for-Polycom-IP500-Ringtones.html
Matt
--
Matt Gibson
Telecommunications
Hi!
I have searched answer how can I transfer calls with asterisk,with no result.
Can you advice me and show some example file how can I use SIP phone to
transfer calls by hitting # and get the Transfer prompt and enter an
extension I want to transfer to?
Thanks for your answers
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure
On Sat, Jul 30, 2005 at 09:51:22PM -0400, Jim Archer wrote:
Hi All...
I'm trying to use the record() app and it complains that it can't open it's
file because permission was denied. I'm running the released Asterisk on
Debian Linux. The target directory is workd writable. Here is the
On Thursday 28 July 2005 18:28, snacktime wrote:
On 7/28/05, wassim darwish [EMAIL PROTECTED]
wrote:
what is the most stable linux that we can build
business on it, i mean the best linux a linux without
problems .
I have Suse 9.1. I had no problems installing it. It is not
the latest
Hi:
I used astcc to create database. After I get the
message database created, I save the configuration and
I move to the next step to assign trunk and route. But
I get the message:
Database unavailable -- please check configuration
Cannot edit routes until database is configured
I checked the
You'll have a much more flexible solution if you keep your MySQL access out of
the * dialplan, and put it in AGI.
Matthew Boehm wrote:
What the hell? NO!
show application MySql
app_addon_mysql is the name of the module.
load app_addon_mysql.so
-Matthew
Quoting Ronald Wiplinger [EMAIL
Hello everybody,
now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to
use PostreSQL instead of MySQL?
Regards
Bastian
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Commercial plug.
Signate is the North American distributor for XC-AST, call queue monitoring
and reporting software for Asterisk. It allows managers to monitor queues
and agents in real time, or to analyze queue activity for given periods.
Real time facilities allow managers to monitor:
No. It's not that. I know that was a problem previously, but I've had
the same problem as the user mentioned and those emails aren't in my
Spam folder. It's like they've completely disappeared.
I guess that's why they call it Beta. :-)
On 8/1/05, Time Bandit [EMAIL PROTECTED] wrote:
I have
--- Time Bandit [EMAIL PROTECTED] wrote:
I have not been receiving mail from the list 29th
July, what is the problem
with gmail and the list.
No problem here.
Mine stopped on the same data, July 29. I had to
subscribe as a new account to get mail from the list.
I checked the log of my
Huddleston, Robert wrote:
Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
___
Asterisk-Users
Time Bandit wrote:
I have not been receiving mail from the list 29th July, what is the problem
with gmail and the list.
Suddenly as is well and I am getting mail again on my normal account.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
I have not been receiving mail from the list 29th July, what is the problem
with gmail and the list.
No problem here.
Check you Spam folder, and if you find email there from this list,
select them all and click Not spam
The list server took a dump last week and has been off line since
I got 23 emails since friday. And im NOT using gmail.
Kyle
Time Bandit wrote:
I have not been receiving mail from the list 29th July, what is the problem
with gmail and the list.
No problem here.
Check you Spam folder, and if you find email there from this list,
select them all and
Huddleston, Robert wrote:
Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
Apparently, a LOT did. Including myself.
Doug
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Hall, Eric M. wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything else but
no luck on the agent part..
I can share mine.
Shows a list of
1: Upgrade your GXP to the latest firmware. See www.grandstream.com
2: [line1] number [send] speak [hold] [line2] number [send] speak
[transfer]
--Rob
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Same here...
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Monday, August 01, 2005 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] test message - ignore me
Haven't seen email
well over 10,000 users getting 80+ emails a day, it was bound to go
down. I wonder how this ranks in the size of mailing lists. Other
than LKML what other lists would be this size?
On 8/1/05, Matt Hess [EMAIL PROTECTED] wrote:
Haven't seen email since the 29th.. just testing.
Me neither.. but just started receiving now. WEIRD.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess
Sent: Monday, August 01, 2005 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] test message -
I have no spam lists. :P
It died for many people I know.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, August 01, 2005 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] what is
Hello everyone,
Where can I find instructions on how to install PHPAGI?
BTW, what's the difference between PHPAGI and PHPAGI2? Are they
different products? It's hard to tell from voip-info.org...
Best,
Leo
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On Mon, Aug 01, 2005 at 02:46:55PM -0400, Huddleston, Robert wrote:
Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
See the IRC channel. The list has been broken for a couple days. If you look
at the archives on
Nope, I have the same problem, nothing.
I jumped on the ISP for not being able to get my mail. Ooops.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Huddleston, Robert
Sent: Monday, August 01, 2005 2:47 PM
To: 'Asterisk Users Mailing List -
Nope, I have the same problem, nothing.
I jumped on the ISP for not being able to get my mail. Ooops.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Huddleston, Robert
Sent: Monday, August 01, 2005 2:47 PM
To: 'Asterisk Users Mailing List -
I have a TDM400P with one FXS and one FXO..
how many liscence(2) I will have to buy?
Been discussed several times on the list (check the archives) and
on the wiki.
Essentially, need a license for each codec translation, including
gsm sounds - g729, etc.
The TDM card does not support g729,
On Monday 01 August 2005 14:53, Innocent Evil wrote:
I have a TDM400P with one FXS and one FXO..
how many liscence(2) I will have to buy?
The licenses don't work that way.
If Asterisk has to rip apart or assemble a g729 stream for any reason, you'll
need a license to do so. If you need to do
I'm running a recent CVS build under Solaris 10.
In the shell than I'm running the Asterisk console I have TZ=US/Eastern
and in my voicemail.conf I have tz=eastern and
eastern=America/New_York|'vm-received' Q 'digits/at' IMp.
The voicemail envelope information seems to be exactly 4 hours
I'm not on gmail, and also haven't received messages since 7/29 -- just
now beginning to see a few trickle in.
-Original Message-
From: Time Bandit [mailto:[EMAIL PROTECTED]
Sent: Monday, August 01, 2005 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I received an off-list follow-up to this, so I figured I'd post some
more info about how I got it to work:
exten = _91XXX, 1, Voicemail(u${EXTEN:2})
exten = _91XXX, 2, HasNewVoiceMail(${EXTEN:2})
exten = _91XXX, 3, Hangup
exten = _91XXX, 103, System(sed 's/__EXTEN__/${EXTEN:2}/'
There are 2 methods blind and announced here you go:
Blind:Call someone, or receive a call. Hit 'Trnf'
The screen displays TRANSFER TO? and you hear a dial tone.
The other end can still hear you, so don't say anything nasty.
Dial the number and hit 'Send', caller is transferred (blind)
the flash operator panel does it great.
You need to use the agent channels instead
of the zap/sip channels.
Try this:
[Agent/101]
Position=3 ; Button number in the console
Label=Steve 101
Extension=101; Extension to reach that channel
Context=localext ; Context where
I currently have two channel groups in my zapata.conf file. I would like
one group to be immediate=yes and the other immediate=no
Does not seem to matter which way I go, the first entry in overrides my
explicit setting for the second group. I am running * 1.0.9 on FC1
[trunkgroups]
On Mon, 2005-08-01 at 10:53 -0800, Innocent Evil wrote:
I have a TDM400P with one FXS and one FXO..
how many liscence(2) I will have to buy?
Short answer: None.
Long answer: Zap interfaces use G711 and do not need G729 to work.
Only if you plan to connect SIP or IAX phones
I do not think the problem with the lists since sometime July 29th was
specific to Gmail...
If you check the web archives, you'll see both regular posts and
others (non-Gmail) asking about problems.
My best guess is a subscriber's domain expired or some other similar
problem which clogs mailserver
Or you could get it (or at least something similar) for free from
www.asteriskguru.com. A small preview is available here:
http://www.asteriskguru.com/tutorials/queue_stats.html
Its 100% ready, just waiting to be uploaded. (Should be there in the
next few days).
Zoa.
William Boehlke wrote:
Joao Pereira wrote:
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
What is happening is that
Check and make sure that astcc-config.conf is owned by the same process
that owns apache. Usually the problem is that astcc-admin cannot write
to the file due to permission problems.
Darren Wiebe
[EMAIL PROTECTED]
chawki hammoud wrote:
Hi:
I used astcc to create database. After I get the
Rich Adamson wrote:
I have not been receiving mail from the list 29th July, what is the problem with gmail and the list.
No problem here.
Check you Spam folder, and if you find email there from this list,select them all and
click Not spam
The list server took a dump last week
Joseph -
I would love to see something like this if you are willing to share.
Thanks.
Joseph wrote:
Hall, Eric M. wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I
Hi,
I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
having problems with Caller ID. I have run clidtest, and it seems happy
enough, returning:-
server clidtest # ./clidtest /dev/zap/1
Number: 041222, Name: MOBILE
(that number's fake.) However, I'm not getting the
Neither did I.. So I called digium this afternoon and they said they
would have someone look at it..
-Gerard
Huddleston, Robert wrote:
Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
www.footnotess7.com is now open to begin the creation of SS7 that can be
used with asterisk.
Sign up and list which and what parts you would like to work on.
Race V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Friday, July
Hello list,
This sounds interesting. Has anyone looked at the source code of these phone
clients. I would be reluctant to download and install software that could be
a trojan software.
Thanks,
Bill Wesson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
www.footnotess7.com is now open to begin the creation of SS7 that can be
used with asterisk.
Sign up and list which and what parts you would like to work on.
Race V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike M
Sent: Thursday, June 09, 2005
www.footnotess7.com is now open to begin the creation of SS7 that can be
used with asterisk.
Sign up and list which and what parts you would like to work on.
Race V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Sunday,
I have the following problem:
I have installed two T1 digium card (old T100P cards), plus a TDM400 with 4 fxo
modules.
Several times in the week I have thousands of warnings like these in the log
Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG
Aug 1 08:54:47 WARNING[2243]:
You must use the 't' 'T' options in the Dial() command when placing calls to
and from the device.
We had extensions that were combinations of SIP and IAX devices and didn't
want/need this behavior on all of our devices so we setup our extensions
with something as follows:
Exten =
let me know if phpagi is a product, i tought it was just a php class
for programming agi php scripts.
best regards
On 8/1/05, Leo Burd [EMAIL PROTECTED] wrote:
Hello everyone,
Where can I find instructions on how to install PHPAGI?
BTW, what's the difference between PHPAGI and PHPAGI2?
I had the same problem in 1.0.9. We fixed it by moving the [zonemessages]
section above the [general] section so that it gets processed first.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frank
Tarczynski
Sent: Monday, August 01, 2005 6:53 PM
I've been using realtime to store my voicemail configuration in a mysql
table for several months now, and have had no problems...until today. A
few weeks ago, I upgraded to the latest CVS and today I noticed voicemail
is not updating the password when the user changes it through option 0.
I'm not
i have a very problem , how to configure MFC/R2
with asterisk, I'am install o module but while asterisk loaded is module is
broken
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Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO.
I've been able to work the FXO ports out and been able to make and
receive calls using softtel PC phones. I'm having difficulty with
configuring 4 line non-PBX analogs to function on the FXS side tho..
I've
Er, make that TDM400P cards... X.X
rc
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Most of the documentation I have read through shows dial plan examples that
dial the SIP phones and stop if one is picked up. I have not seen an example
of or read how to stop the SIP dial when an analog phone is answered.
How can the extension be set up so that when an analog phone is picked up
Hello,
I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.
1.I couldn't find Asterisk version using asterisk -V command.
How can I to find version information?
2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on
it.
I
The clock on cisco phones 'disappears' when it fails to receive
updates from the ntp server.
This is most likely due to your ntp server configuration. By default
the ntp mode on your cisco phone is directedbroadcast. If your ntp
server doesn't support this you will need to change the mode on
Anybody using Cisco Call Manager and connecting to any SIP termination
service like voipjet, voxee, etc? Please msg me offlist.
AK
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If the SC420 is sharing interrupts, can you go around that by chaning slots
or maybe, I don't know if it can do APIC? Or how about disabled the shared
devie like USB?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Joe McConnaughey
|Sent: Lunes, 25
Hello everyone, I have just received 3 brand new Polycom SoundPoint IP
600 from voisupply.com and I have the exact same problem on all of
them. When I receive a call, the phone is ringing correctly but when I
answer it, it takes exactly 10 seconds before I can hear the caller. I
also have
You are duly ignored.
Matt Hess wrote:
Haven't seen email since the 29th.. just testing.
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Thanks everybody for answering me.
Yes, I plan to connect SIP phone outside of my network. Infact, I am going
to use Asterisk as my PSTN gateway and voice mailbox.
Also, I have plan to add two more FXO card when I will have bigger network.
Sounds like, I should get two liscences at this moment.
Please look on the [EMAIL PROTECTED] Fourms
On Mon, 1 Aug 2005, Robert Chapin wrote:
Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO. I've
been able to work the FXO ports out and been able to make and receive calls
using softtel PC phones. I'm having
Have you used the automatic configuration script for the zaptel drivers?
IF so, have you added ZAP extensions in AMP for your analog phones?
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Chapin
Sent: Monday, August 01, 2005 9:13 PM
To:
I may be mistaken, but in [EMAIL PROTECTED], can't you just press # and dial the
extension number , speak, and hang up?
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phoneguy
Sent: Monday, August 01, 2005 7:02 PM
To: Asterisk Users Mailing
Yes and yes. Zap Trunks and Extensions were added for the analogs.
rc
Tom Rymes wrote:
Have you used the automatic configuration script for the zaptel drivers?
IF so, have you added ZAP extensions in AMP for your analog phones?
Tom
-Original Message-
From: [EMAIL PROTECTED]
We will start installing TE411 next week, I'll keep the list informed !
jack
Eric Rees wrote:
Has anyone on the list tried one of these new cards with built-in echo
cancellation?
This electronic message transmission, including attachments, is for the
exclusive use of the individuals to
Hi
We purchased the AT320-EE IAXtalk phone from www.iaxtalk.com which ocnnects to
our own asterisk server.
Good value, a little tricky to set up - the instructions they supply to which
they give you a link on their web site are OK, but their are some gaps which
the asterisk wiki pages fill
The REV I card shows up in the PCI table as:
02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or
02:05.0 Class 0280: e159:0001)
Subsystem: Unknown device b119:0001
But the REV E/F shows up as:
02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
I have just installed the TE4110P card, found no real issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of pbx
Sent: Tuesday, August 02, 2005 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] New digium
On Tue, Aug 02, 2005 at 02:48:00PM +1000, Edwin Groothuis wrote:
The REV I card shows up in the PCI table as:
02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or
02:05.0 Class 0280: e159:0001)
Subsystem: Unknown device b119:0001
But the REV E/F shows up as:
And where did you get your rate?
The 11/2004 rates from nufone show:
Taiwan 886 0.0469
Taiwan - Mobile/Special Services886 60 0.1006
Taiwan - Mobile/Special Services886 70 0.1006
Taiwan - Mobile/Special Services886 9 0.1006
Taiwan -
On 8/2/05, John Novack [EMAIL PROTECTED] wrote:
Rich Adamson wrote:
I have not been receiving mail from the list 29th July, what is the
problem with gmail and the list.
I suspect that any dump the list might or might not have taken isn't
the complete story.
m using gmail and
I lost a few too.
I jumped from Vol. 12 Issue 199 to Vol. 13 Issue 3.
Anybody know exactly how many issues to a volume? - I've seen it vary quite
a bit (i.e. 208 issues in Vol. 11 / 268 in vol. 9).
Would be kinda' nice if they'd pick a number (like maybe 200?) and stick
with it???
If
On Mon, Aug 01, 2005 at 11:56:26PM -0500, [EMAIL PROTECTED] wrote:
On Tue, Aug 02, 2005 at 02:48:00PM +1000, Edwin Groothuis wrote:
The REV I card shows up in the PCI table as:
02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or
02:05.0 Class 0280: e159:0001)
I have cvs-head of Aug-2. README has no information on how to bind
asterisk-h323 on multiple interfaces. actually this was my question that
can we bind asterisk-h323 on multiple interfaces ? as h323.conf says
that bindaddr should contain a single valid IP.
if we bind h323 to 0.0.0.0 as
hello
is there any way to register all user without
declaring them in sip.conf. because i want all users
to auth.
thanks in advance
Kamran
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything else but
no luck on the agent part..
How are your agents loging into queues? Depending on that
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