Hi all,
Okay I have solved the problem.
Actually the asterisk detected 24 Port FXO and numbered its ports. Since
it has previously detcted ports 1-4 for FXS and ports 58 for FXO for my
initial 8-port card. When I installed 24 port second card it numbered
the new fxo ports from 9-32. uptill now
On Mon, Aug 04, 2008 at 11:34:06AM +0500, Syed Nasruddin wrote:
Hi,
Can anyone help me on this. I am really stuck.again defining the problem
briefly.:
1. Second New card TDM240P added to machine.
2. Only FXO modules i.e 24 FXO.
3. Asterisk detected all the ports successfully and
Hi to all
except of some commercial hardware / software gateways, is there any
opensource or free project to setup a Skype Account on Asterisk?
Thanks to all
--
/*/
nik600
http://www.kumbe.it
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Hi All,
I have this Call Center requirement to be implenetd:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5 agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Kindly suggest
Hi:
I want to asterisk play gsm files instead mp3 files when only one
person is in a conference room with 'M' option in Meetme application.Is
it possible?
(I place 2 gsm files in mohmp3 folder and didn't install mpg123)
I'd appreciate any help.
Alex Balashov a écrit :
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5 agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is (as
mentioned in below example) if a call enters testq and get answered then
after hungup at the agent end only will the call will again enter the
next queue which is testq2 as in this example.??
Sorry for previous blank answer :)
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is (as
mentioned in below example) if a call enters testq and get answered then
after hungup at
On Mon, Aug 4, 2008 at 10:58 AM, nik600 [EMAIL PROTECTED] wrote:
Hi to all
except of some commercial hardware / software gateways, is there any
opensource or free project to setup a Skype Account on Asterisk?
The only one known to the moment is chan_celliax, which is originally
for connecting
Syed Nasruddin pisze:
Hi All,
I have this Call Center requirement to be implenetd:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5
agents, firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is (as
mentioned in below example) if a call enters testq and get answered then
after hungup at the agent end only will the call
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
--Stefan
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AstriCon 2008 - September 22 - 25 Phoenix,
Hi Stefan,
Stefan Gofferje wrote:
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
I looked into this for my N95 but if it's possible then it isn't
documented. At least I could not find any public documentation how to do
Hi All
I have an AA50 without inbound DDIs but each line has a separate number
so based on analogue port it can be routed to different people. The
challenge with this method is it appears to only allow the dial plan to
use 1 outbound route so if all the analogue ports are split into
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5 agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call Queue() on the first queue -
Dear Atis,
I am running in to syntax problem. Sorry only beginner level experience
of conditional checking:
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup(); since I want to hangup if the caller
has
On Mon, 4 Aug 2008, Patrick wrote:
Hi Stefan,
Stefan Gofferje wrote:
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
I looked into this for my N95 but if it's possible then it isn't
documented. At least I could not
Use penalties and you will be done in two minutes. Try not to reload
while calls are queued.
Thanks,
Steve T
On Mon, Aug 4, 2008 at 7:59 AM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Dear Atis,
I am running in to syntax problem. Sorry only beginner level experience
of conditional
On Sun, Aug 3, 2008 at 6:19 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Curious why you stay with postgres then, and not go with MySQL if you
know in advance it is a problem and will bite you sometime?
The other IT truism, it's not easy to change a production system. In
our case we have a lot of
Hi Everyone,
I have configured Asterisk with MySQL with realtime using sip_buddies and
extensions tables. Everything is ok and connection to mySql DB is fine when I
type the command realtime mysql status on Ast CLI.
I have one entry into sip_buddies table as below.
| id | name |
Gordon Henderson schrieb:
On Mon, 4 Aug 2008, Patrick wrote:
Hi Stefan,
Stefan Gofferje wrote:
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
I looked into this for my N95 but if it's possible then it isn't
Abid Saleem wrote:
Aug 4 15:02:30 NOTICE[17050]: chan_sip.c:11291 handle_request_register:
Registration from '1504sip:[EMAIL PROTECTED]' failed for
'10.168.20.211' - ACL error (permit/deny)
Sounds like Asterisk thinks the corresponding SIP peer does not actually
exist.
--
Alex Balashov
On Mon, Aug 4, 2008 at 2:59 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Dear Atis,
I am running in to syntax problem. Sorry only beginner level experience
of conditional checking:
Yes, sorry for that, i just wrote it quickly and didn't checked
expression. Also, i didn't wrote in .conf
This email is off topic. If it offends you then suck it :-)
But seriously, I know this will interest USA readers (unfortunately for
now primarily on the west coast only). How cool is this concept
http://techshop.ws/index.html
Cheers,
Dean
From: Dean
On Mon, Aug 4, 2008 at 9:24 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Aug 4, 2008 at 2:59 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Dear Atis,
I am running in to syntax problem. Sorry only beginner level experience
of conditional checking:
Yes, sorry for that, i just wrote it
May be. If somebody has experience this problem before, then only he/she can
guide about this. I am not sure whats going on.
Abid Saleem Date: Mon, 4 Aug 2008 09:19:45 -0400 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk
Realtime with MySQL
On Mon, Aug 4, 2008 at 9:34 AM, Dean Collins [EMAIL PROTECTED] wrote:
This email is off topic. If it offends you then suck it J
But seriously, I know this will interest USA readers (unfortunately for now
primarily on the west coast only). How cool is this concept
http://techshop.ws/index.html
On Mon, 2008-08-04 at 16:48 +0300, Abid Saleem wrote:
May be. If somebody has experience this problem before, then only
he/she can guide about this. I am not sure whats going on.
Abid Saleem
Try to set debug verbose option in logger.conf, then check all query
from asterisk to mysql for see
No you've missed the point - it's just for people who want to use the
tools - not meant to be a outsource workshop location.
Cool things like plasma cutters and laser etchers etc.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Thanks Atis and steve.
I think I will have it running tomorrow.
Thanks a lot.
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, August 04, 2008 6:40 PM
To: Asterisk Users Mailing List - Non-Commercial
My problems with bad hdlc,device busy and unable to open D-channel on
span 1 was connected with file indications.conf where I forgot to change
default settings country=us.
Uros
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On Mon, 4 Aug 2008, Stefan Gofferje wrote:
Gordon Henderson schrieb:
On Mon, 4 Aug 2008, Patrick wrote:
Hi Stefan,
Stefan Gofferje wrote:
Hi,
anybody knows if it is possible to make the Nokia SIP client in the
phones autoanswer a call in speakerphone mode?
I looked into this for my
On Sun, Aug 03, 2008 at 07:11:41PM +0100, Femi wrote:
Thanks for the design tips
Based on all the information I have been able to gather this is the config
that I believe will work best:
1. SER on 2 servers in HA (failover) config
2. Asterisk cluster of 4 (or more) servers
3. MySQL and
On Mon, Aug 04, 2008 at 03:45:32PM +0100, Gordon Henderson wrote:
So I'll throw the cat amongst the pigeons by saying: Who cares about PTT?
Me.
Seriously - Push to talk - Half duplex Communications. How ancient is
that! It's really reminiscent of ancient US style truckers - Smokey and
the
Hello all,
I am currently running Ubuntu LTS 8.04.1 Server Edition and I want to build
the Asterisk-1.4.21.2~dfsg-1 (source) from the Debian Unstable repository. I
am wondering on how I can customize or perform the MENUSELECT part when
running the command debuild -us -uc in building the said
On Monday 04 August 2008 07:57:46 Abid Saleem wrote:
| id | name | accountcode | amaflags | callgroup | callerid | canreinvite |
| context | defaultip | dtmfmode | fromuser | fromdomain | fullcontact |
| host | insecure | language | mailbox | md5secret | nat | deny | permit
| | mask |
On Monday 04 August 2008 09:29:21 Dean Collins wrote:
No you've missed the point - it's just for people who want to use the
tools - not meant to be a outsource workshop location.
Cool things like plasma cutters and laser etchers etc.
There are multiple, non-profit clubs in the USA right now,
Hi,
I'm not sure if this is the proper way to approach it but i can't figure out
how to setup dundi.
what i did is, i try to determine which server a user is registered, by calling
an agi to query the realtime db and capture the regserver of the user.
e.g.
exten =
Gordon Henderson schrieb:
Seriously - Push to talk - Half duplex Communications. How ancient is
that! It's really reminiscent of ancient US style truckers - Smokey and
the Bandit and all that. That's just so last century. Lets put all that
behind us and get with the 21st century! We all
On Mon, Aug 04, 2008 at 08:04:31PM +0300, Stefan Gofferje wrote:
If I really want PTT, then I'll go out buy a pair of Motorola handsets.
Which would reach how far? 500m? Surely not from Helsinki to Oulu or
even internationally...
And, as a folo, Nextel is slowly transitioning their Direct
At 9:34 AM -0400 2008/8/4, Dean Collins wrote:
This email is off topic. If it offends you then suck it J
But seriously, I know this will interest USA readers (unfortunately
for now primarily on the west coast only). How cool is this concept
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk 1.4.x and
FreePBX 2.4+. I'm using Polycom phones. I can't make it work. Would
somebody please give a checklist of items for me to compare my list
against - in the
I'm not sure if this is the proper way to approach it but i can't figure out
how to setup dundi.
what i did is, i try to determine which server a user is registered, by
calling an agi to query? the realtime db and capture the regserver of? the
user.
e.g.?
exten =
On Mon, Aug 4, 2008 at 2:18 PM, JR Richardson [EMAIL PROTECTED] wrote:
Use DUNDi, perfect for this. The protocol is very light, no load on
the servers to run it, can handle hundreds of queries a second with no
load. You want to use regcontext and a few other things to make it
all work
On Mon, Aug 04, 2008 at 11:13:30PM +0800, GNUbie wrote:
Hello all,
I am currently running Ubuntu LTS 8.04.1 Server Edition and I want to build
the Asterisk-1.4.21.2~dfsg-1 (source) from the Debian Unstable repository. I
am wondering on how I can customize or perform the MENUSELECT part when
- Elliot Murdock [EMAIL PROTECTED] wrote:
Thanks Tzafrir,
This is what I get:
module unload chan_zap.so
-- Unregistered channel -2
-- Unregistered channel 1
...
-- Unregistered channel 122
-- Unregistered channel 123
-- Unregistered channel 124
CLI module
On Mon, Aug 4, 2008 at 1:30 PM, John Todd [EMAIL PROTECTED] wrote:
At 9:34 AM -0400 2008/8/4, Dean Collins wrote:
This email is off topic. If it offends you then suck it J
But seriously, I know this will interest USA readers (unfortunately for now
primarily on the west coast only). How cool
I installed a new machine with CentOS 5.2, Zaptel 1.4.11 and Asterisk
1.4.21.2 and an OpenVox A1200P card. This card has its own driver and
Zaptel has been patched to use it. The problem is that from the moment
I load Zaptel I get this messages on the console:
buffer re-sync occur from
The only one known to the moment is chan_celliax, which is originally
for connecting to cell phones by cable, however it supports also skype
(just 1 account). It will launch fake X server and original skype, and
communicate with it.
http://www.celliax.org/
Free: sippyskype (same quality as
Hi,
I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist
--
Regards,
Sahil Gupta
Corporate Advisor
TigerCom Pte. Limited
296 River Valley Road
Singapore 238337
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I'm seeking an 8 port FXO gateway. Please let me know if anyone can assist
Use google and such, or at least specify the location
(europe,usa,australia,whatever)
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AstriCon 2008 -
I suppose the bit to check is the features ('show features') and then
try to record a call (*1) and see what the terminal says...
PaulH
Bill Michaelson wrote:
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk
Hi,
A closer look showed that SIP FOP-Originated calls are self-addressed
While some phones tolerate that, others reply with 480 moved temporarily.
Case 1: Command Line Interface with Thomson hardphone
After I typed originate SIP/9122 application dial Local/[EMAIL PROTECTED], 1st
SIP message
Hi Everyone,
Those of you who have a simple home-based Asterisk box might
be interested in a simple Win32 (Win2K or WinXP) interface to
the AMI manager. The quick-start versions merely require
unzipping with NO Installation - hence, NO Uninstall (i.e., no
registry writes at any time by the
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