As a dual GSM/WiFi mode phone might be already engaged in a GSM conversion
while a SIP call occurs, I think Symbian application dev rules would impose
any application to centralize microphone and speaker allocation to a Symbian
provided resource manager.
So I think automatic answer of any kind
Hi Everyone,
I am currently running Trixbox 2.6 and I have a problem with Asterisk.
/etc/init.d/asterisk status
Asterisk dead but subsys locked
I deleted all files in /var/run/asterisk folder and asterisk restart...
It's ok for a while. But some days after Asterisk again is dead.
Can anybody
Dear Mr. Tilghman,
Thank you for your attention. Actually it was NULL before when it was not
working. I changed it to deny=no and permit=all after that thinking it could be
the problem. Now I have changed it back to NULL using update sip_buddies set
deny=NULL, permit=NULL where id=1. You can
Steven Howes wrote:
On 5 Aug 2008, at 09:16, Budacsik Attila wrote:
Hi Everyone,
I am currently running Trixbox 2.6 and I have a problem with Asterisk.
/etc/init.d/asterisk status
Asterisk dead but subsys locked
I deleted all files in /var/run/asterisk folder and asterisk
restart...
Olivier schrieb:
Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to
a SIP call.
Is it certain ?
Yes, just tested it myself. Phone answers with Busy here if in a GSM call.
From my understanding, Symbian applications MUST leave this decision
type to an external
2008/8/5 Stefan Gofferje [EMAIL PROTECTED]
Olivier schrieb:
As a dual GSM/WiFi mode phone might be already engaged in a GSM
conversion while a SIP call occurs, I think Symbian application dev
rules would impose any application to centralize microphone and speaker
allocation to a Symbian
On 5 Aug 2008, at 09:16, Budacsik Attila wrote:
Hi Everyone,
I am currently running Trixbox 2.6 and I have a problem with Asterisk.
/etc/init.d/asterisk status
Asterisk dead but subsys locked
I deleted all files in /var/run/asterisk folder and asterisk
restart...
It's ok for a while.
Olivier schrieb:
As a dual GSM/WiFi mode phone might be already engaged in a GSM
conversion while a SIP call occurs, I think Symbian application dev
rules would impose any application to centralize microphone and speaker
allocation to a Symbian provided resource manager.
So I think
Hi,
I am using Asterisk 1.4.18. I am implementing Penalties for my agents.
What is happening: two agents configuired one agent with penalty 1 and
the other with penalty 2. All the calls must go first to Agent 1 and if
his line is busy then only then agent 2 will get the call. However my
queues
Steven Howes wrote:
On 5 Aug 2008, at 09:16, Budacsik Attila wrote:
Hi Everyone,
I am currently running Trixbox 2.6 and I have a problem with Asterisk.
/etc/init.d/asterisk status
Asterisk dead but subsys locked
I deleted all files in /var/run/asterisk folder and asterisk
restart...
You're certainly right :
today, it's possible to insert a GSM PCI card in a PC, load it with Asterisk
and do whatever you want with incoming and outgoing calls but the fact is,
AFAIK, we can't do the same with a handset (you (mostly) can't edit Least
Cost Routing rules, you can't autoanswer, etc
Olivier schrieb:
Which company publishes this Symbian application implementing a local
answering machine on the phone, for instance ?
There are several. For instance, rock your mobile comes to my mind.
http://www.rock-your-mobile.com/
http://www.rock-your-mobile.com/answering-machine.php
-S
Syed Nasruddin wrote:
Hi,
I am using Asterisk 1.4.18. I am implementing Penalties for my agents.
What is happening: two agents configuired one agent with penalty 1 and
the other with penalty 2. All the calls must go first to Agent 1 and
if his line is busy then only then agent 2 will get
Hello,
Just wanted to let you know that the XP version works fine on vista.
I was working on a similar program but didnt have enough time to finish, I
was working on Delphi 7 btw.
Thanks
Marco.
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Gerald
On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez
[EMAIL PROTECTED] wrote:
Syed Nasruddin wrote:
Hi,
I am using Asterisk 1.4.18. I am implementing Penalties for my agents.
What is happening: two agents configuired one agent with penalty 1 and
the other with penalty 2. All the calls must go
You have to limit calls to these agents, use incomminglimit or call-limit on
sip.conf to do that.
That way, when the first agent answers a call, all the other calls directed
to it will return with busy signal, and will be transferred to the other
agent.
__
Marco
Cannot i use ringall strategy with penalties???
Will rrmemory will fullfil my requirement??
My requirements:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5 agents,
firstly.
3. IF ALL the first 5 agents are busy then ONLY then the call will be
On Tue, Aug 5, 2008 at 1:55 AM, Gerald Harshany [EMAIL PROTECTED] wrote:
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
On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Cannot i use ringall strategy with penalties???
Will rrmemory will fullfil my requirement??
rrmemory isn't ringall, it won't ring all members. But yes - you can
use ringall with penalties.
My requirements:
1. 10
I realize this may be slightly off-topic but I'm wondering if someone here can
lend me a hand.
One of my GXW4008 has gone unconfigurable via standard HTTP (refuses
connection) and I can't use the built-in IVR because I had previously disabled
the keypad update feature. So I'm stuck with just
When calling from our SIP proxy through Asterisk to the PSTN provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP
media path. I understand that
Never mind. I set the wrong baud rate. The right values are 115200-8-N-1-No
flow control. However, the serial connection is as good or as useless as the
telent connection. I have no way to restore factory settings.
--- On Tue, 8/5/08, Vieri [EMAIL PROTECTED] wrote:
I realize this may be
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original
HI folks! my topology is:
softswitch (BROADSOFT) -- [sip trunk] -- Asterisk
I need to connect phone calls using g729 codec. Debugging some calls we
found that calls cant connect because of codec incompatibility. Our Sip
provider send us annexb=yes when a call is comming
I don't think you can do that because, asterisk, in the caller thread
will only read MACRO_RESULT to know if he has to connect the call or not.
A workaround will be to :
1. before the dial, add a row in a database table and retrieve an id
2. pass the id to test_connect and test_connect will then
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable
Dear people, does anybody try the ZRTP patch for Asterisk in order to
have ZRTP encrytion among SIP/RTP calls ???
In other words, did anybody succesfully implement ZRTP in Asterisk ???
Any documentation about it ???
Special thanks
Alejandro
___
--
And if you use DIALSTATUS and ANSWERTIME to check the last dial status,
you need to take care of the following bug
http://bugs.digium.com/view.php?id=13216
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL
Hi,
Actully the way I want the penalties functionality to behave it is not
doing it accordingly. I am right now using ringall. Set penalty 1 for
one agent and 2 for secnd agent. All the calls come in and go to first
agent#1 having penalty one. But the second call also go to agent#1 and
start
I would suggest putting a NOOPIn the MACRO to ensure the variable IS
actually getting SET.
As I understand it VARIABLES are GLOBAL and what you are doing is
correct, BUT, This could be a learning opportunity for me too.
Be advised, there seems to be push by DIGIUM for folks to use the
On Tuesday 05 August 2008 03:49:15 Abid Saleem wrote:
Dear Mr. Tilghman,
Thank you for your attention. Actually it was NULL before when it was not
working. I changed it to deny=no and permit=all after that thinking it
could be the problem. Now I have changed it back to NULL using update
Err - Ok - let me ask this in MUCH simpler way
1 - In dialplan , you set a Variable called MYVAR, to Apple
2 - You go into MACRO and NOOP the VALUE of MYVAR -What SHOULD it BE ?
3 - While IN MACRO you set VALUE of MYVAR = Pear
4 - You leave MACRO and get back to DIALPLAN and NOOP the value
Syed Nasruddin wrote:
Hi,
Actully the way I want the penalties functionality to behave it is not
doing it accordingly. I am right now using ringall. Set penalty 1 for
one agent and 2 for secnd agent. All the calls come in and go to first
agent#1 having penalty one. But the second call also
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk) TO the avaya
box, and it accepts that and sets
above the original post is very confusing. Please stop doing this.
The format of this post is in reverse, to demonstrate why posting a reply
option is only in trunk. So no, it would not help him out.
Yes, it works the same way, by using the U() option to Dial. However, this
Question 2:
prior
On Tuesday 05 August 2008 18:04, Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp(
On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote:
I don't think you can do that because, asterisk, in the caller thread
will only read MACRO_RESULT to know if he has to connect the call or not.
A workaround will be to :
1. before the dial, add a row in a database table and retrieve an id
On Tuesday 05 August 2008 18:50, Al Baker wrote:
Err - Ok - let me ask this in MUCH simpler way
1 - In dialplan , you set a Variable called MYVAR, to Apple
2 - You go into MACRO and NOOP the VALUE of MYVAR -What SHOULD it BE ?
You should better use
M(x[^arg]) - Execute the Macro for
On Tuesday 05 August 2008 14:19:44 Thomas Winter wrote:
On Tuesday 05 August 2008 18:04, Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL
Hello,
When sending this AMI request ...
192.168.64.5 - Action: Originate
192.168.64.5 - Channel: SIP/9122
192.168.64.5 - Async: True
192.168.64.5 - Callerid: 9122 Guest2 9122
192.168.64.5 - Exten: 9123
192.168.64.5 - Context: local
192.168.64.5 - Priority: 1
... I've got this INVITE from
Thomas Winter wrote:
On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote:
I don't think you can do that because, asterisk, in the caller thread
will only read MACRO_RESULT to know if he has to connect the call or not.
A workaround will be to :
1. before the dial, add a row in a
All,
I have a problem. The company I work for has been bought out by a bigger
company. The employees are in the process of changing all their email
addresses to the new company name. I have my voicemail.conf file setup
to email users when they have a voicemail message. The mail server that
Brian Simpson wrote:
All,
I have a problem. The company I work for has been bought out by a bigger
company. The employees are in the process of changing all their email
addresses to the new company name. I have my voicemail.conf file setup
to email users when they have a voicemail
On Tue, Aug 5, 2008 at 3:10 PM, Brian Simpson
[EMAIL PROTECTED] wrote:
the network that the Asterisk sits on. I have change a couple to test
but the email notification is not happening. Any idea what is going on
Is sendmail installed and running?
___
Hi Brian,
Interesting company you work for :) - would be good to see some
Asterisk integration into the azure network.
Are you sure that the Asterisk server has SMTP access to the outside
world?
Just because you are changing the ip address of the server it was being
sent to originally to
On Tuesday 05 August 2008 17:10:17 Brian Simpson wrote:
I have a problem. The company I work for has been bought out by a bigger
company. The employees are in the process of changing all their email
addresses to the new company name. I have my voicemail.conf file setup
to email users when they
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Tuesday, August 05, 2008 6:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] email notification to external email address
Brian Simpson wrote:
The Asterisk development team has released new versions of three
libraries used with Asterisk. They are:
libpri-1.2.8:
This release contains a number of bugfixes that had been unreleased for
months, along with clarification of the licensing of the source code.
The change log is here:
Brian Simpson wrote:
All,
I have a problem. The company I work for has been bought out by a bigger
company. The employees are in the process of changing all their email
addresses to the new company name. I have my voicemail.conf file setup
to email users when they have a voicemail message.
Syed Nasruddin wrote:
Actully the way I want the penalties functionality to behave it is not
doing it accordingly. I am right now using ringall. Set penalty 1 for
one agent and 2 for secnd agent. All the calls come in and go to first
agent#1 having penalty one. But the second call also go to
Does anyone know where I might purchase a G.722 capable SIP soft phone?
Counterpath say that they offer one, but only in the OEM versions do
they support G.722. I need only a couple of licenses.
Thanks,
Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN
On Tue, Aug 5, 2008 at 6:29 PM, Vieri [EMAIL PROTECTED] wrote:
Never mind. I set the wrong baud rate. The right values are 115200-8-N-1-No
flow control. However, the serial connection is as good or as useless as the
telent connection. I have no way to restore factory settings.
--- On Tue,
ASTPP (www.astpp.org) will do calling cards / prepaids as well as lcr.
Darren Wiebe
[EMAIL PROTECTED]
emist wrote:
Hello,
does anyone know of a good calling card solution for asterisk that is
able to do lcr?
Does astcc do this? I've been searching around and I can find some lcr
On Tue, Aug 05, 2008 at 07:41:05PM -0400, Andres wrote:
Brian Simpson wrote:
All,
I have a problem. The company I work for has been bought out by a bigger
company. The employees are in the process of changing all their email
addresses to the new company name. I have my voicemail.conf
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