I have the same problem.
callprogress is very experimental and buggy now.
and i've lost the .call files feature of asterisk.
what do you think about submitting a bug on bugs.digium.com?
regards,
shabanip
Hello,
I've been experimenting with the call progress analysis features of *,
with mixed
see: http://www.voip-info.org/wiki-Asterisk+billing
- shabanip
- Original Message -
From: Thomas Kuepper [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 12:42 PM
Subject: [Asterisk-Users] CDR Dokumentation
ist there any cdr dokumentaion about the cdr format?
thx
also: http://www.voip-info.org/wiki-Asterisk+cdr+csv
- Original Message -
From: Thomas Kuepper [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 25, 2004 12:42 PM
Subject: [Asterisk-Users] CDR Dokumentation
ist there any cdr dokumentaion about the cdr format?
thx!
thomas
is there any way to read global vars like ${EXTEN}, ${GROUPCOUNT} from an
AGI?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
is it possible to windows messenger clients of an asterisk server to chat
(text chat) with each other?
what about the status presence? is it possible to each windows messenger
client of an asterisk server to see the presence on other clients?
if not, what is missing in asterisk?
are not global vars.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of shabanip
Sent: Monday, October 11, 2004 10:17 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] reading global vars from AGI
is there any way to read global vars like ${EXTEN
]: chan_zap.c:5903 do_monitor: Whoa
I'm owned but found (26)...
Sep 26 10:14:26 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa
I'm owned but found (24)...
Don't cross messages between lists.
Anyway, be more specific.
- Original Message -
From: shabanip
To: [EMAIL
I have the same problem and
suppose that by some works on a new application similar to
parkandannounce app. it should be done.
- shabanip
- Original Message -
From: Alex Forrow [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 17, 2004 5:43 PM
Subject: [Asterisk-Users
Title: Message
see: http://www.voip-info.org/wiki-Asterisk+billing
- Original Message -
From:
Mayank Mishra
To: [EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 2:10
PM
Subject: [Asterisk-Users] How to get Call
Details Records
HI,
Can anyone please
I get this message at CLI.
what does it mean?
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
what does it do?
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Monday, September 06, 2004 1:33 AM
Subject: [Asterisk-Users] res_perl
Latest version of res_perl is up also.
a call
to an extension, there's no way to have the caller transfered back to
yourself if the called extension doesn't answer or if it's busy. is this
correct?
thanks,
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
a call
to an extension, there's no way to have the caller transfered back to
yourself if the called extension doesn't answer or if it's busy. is this
correct?
thanks,
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
a call
to an extension, there's no way to have the caller transfered back to
yourself if the called extension doesn't answer or if it's busy. is this
correct?
thanks,
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com
Is there any way to play a sound
during a call between two endpoints?
-shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
ofcourse the sound should be heared by two sides of the call?
- shabanip
Is there any way to play a sound
during a call between two endpoints?
-shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
I've found an incorrect timezone in GS firmware:
Tehran timezone is +3:30 not +3:00
It gets definitely better every day.
List of bug fixes follows:
Release 1.0.5.9 7/26/2004
If SIPRegister doesn't proceed due to conditions unmet, release
channel resource
Fix the LED
I see my own number(or remote called num) instead of caller id on incoming
calls on my BT-102.
but on Xlite everyything is OK. I'm using * latest CVS.
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman
: shabanip [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 26, 2004 8:40 AM
Subject: [Asterisk-Users] GrandStream CallerID
I see my own number(or remote called num) instead of caller id on
incoming
calls on my BT-102.
but on Xlite everyything is OK. I'm using * latest CVS.
- shabanip
- this is my problem too. I couldn't find a reseller which could send me
one in a reasonable time.
- shabanip
- Original Message -
From: Jan Goericke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 16, 2004 1:32 PM
Subject: [Asterisk-Users] some questions on uniden uip200
hello
Uniden-200
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 12:17 PM
Subject: [Asterisk-Users] SIP phones recommendations
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all.
We are currently using either Grandstream
Uniden UIP-200
you can follow this thread in the list: Cheap (US$120 or less) SIP Phones
- Original Message -
From: shabanip [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 1:21 PM
Subject: Re: [Asterisk-Users] SIP phones recommendations
Uniden-200
- Original
I have problem in configuring MP124 FXS Gateway to work with *.
Can anaybody help me in this way?
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
I have problem in configuring MP124 FXS Gateway to work with *.
Can anaybody help me in this way?
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
4.7 to 5.0.0482.
- shabanip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I have the same problem but with MP-124 FXS Gateway.
Does anybody has it to work with *?
- Original Message -
From: Brian J. Rathman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 6:52 PM
Subject: [Asterisk-Users] Audiocodes - Asterisk Implementation
Anyone out
I'm using version 1.9.1 build 3908
- next problem is that the text messages won't reach by another firefly
client
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
I'm using version 1.9.1 build 3908
- next problem is that the text messages won't reach by another firefly
client
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Randy Bush [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday, June 29, 2004 7:31 PM
Subject:
- Would you please give me a sample config params in sip.conf
or where ever it should be?
- where can i download old versions of MSN messenger?
Hi,
-Original Message-
Is there any software based solution to establish a video
connection with * and sip protocol?
MSN messenger 4.7
Is there any software based solution to establish a video connection
with * and sip protocol?
- Original Message -
Hi,
-Original Message-
It's already possible to use VideoPhone with Asterisk.
I'm planning to buy 2 of them. Anybody using any Video SIP
phone with
Is there any way to handle
CDRswith AGI?
so my question changes to:
- How to create a CDR backend?
- I want to run my own scripts.
shabanip
shabanip wrote:
Is there any way to handle CDRs with AGI?
There is no need to. Install or create a CDR backend.
Jeremy McNamara
- voicemail, ACD, IVR and MOH will be used
- conference will be rarely used.
whats your recommended pc now? Is one * box enough for this job?
shabanip
- Original Message -
From: Anton Tinchev [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 04, 2004 7:07 AM
Subject: Re
- conference will be rarely used.
whats your recommended pc? a quad xeon or even a quad opteron with xxGB
of memory? is it enough for this big job?
shabanip
.G,
Come on now, give us more.
How many concurrent calls? What's your idea of a modern PC? Processor
Speed, HardDrive space, etc
I'm using * on 3Ghz P4 with HT enabled with a TE405P card with no problem.
I'm using fedora 2 but made to change the kernel to 2.6.6.
- Original Message -
From: Chris Bond [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 4:50 PM
Subject: RE: [Asterisk-Users] Re:
anybody has success stories about running * on AMD
Opteron?
Use r option in your Dial command.
- Original Message -
From: joachim [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 2:45 AM
Subject: Re: [Asterisk-Users] No ringing sound on GS phones
Make sure to use CVS-head and you'll get ringing.
At 23:51 28/05/2004, you
Title: Max TE410P card on an Asterisk
Hello,
Does
anybody know the max number of TE410P/TE405P card we can put in an asterisk
box?
Thanks.
38 matches
Mail list logo
