Thanks Adam, replies inline:
Adam Goryachev wrote:
On Thu, 2005-07-21 at 15:30 +0100, Asterisk wrote:
I've got several agents on a queue. However, they often forget to go
not ready or log off when they can't answer the phone.
I would like a person calling my queue to be on the queue for a
Hello,
Just chiming in here:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of
[EMAIL PROTECTED]
Sent: Viernes, 22 de Julio de 2005 01:24 p.m.
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Dell Hardware
Mmhh nice !! So, why did Digium
On Sat, 23 Jul 2005, Razza wrote:
All,
before I start, I have read the readme and realise the dial syntax has
changed since 3.5.(and I really wish it hadn't) anyway can
somone please help me round a config issue?
Previously I had a capi.caonf as follows. This config allowed me to
On Fri, 2005-07-22 at 15:42 -0500, Eric Wieling aka ManxPower wrote:
Eric Rees wrote:
We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX
sounds better then the land line.
Using UDP or using TCP?
Dear frinends,
I am useing the asteirsk for ACD features my dobut is this in queues.conf
we can give the announce hold time yer or no .
If we give how yes how can we say that this hold time dynamically to the queuemembers .
are we manually setting any where that this call need to be last only for
Hi,
asterisk compiled fine and is running very stable on
our dual opteron in 64 bit mode.
When loading G.729 library we have to peload libz manually for any
reason, but besides that minor issue, everthing is fine.
We didn't yet test the limits of that machine.
Roger.
U can try the firefly. this softphone can be used w/ more then 1 line...
https://www.virbiage.com/download.php
To configurate this in asterisk is like a normal sip phone or iax phone...
-SIP.CONF
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and
On Fri, Jul 22, 2005 at 10:56:42AM -0500, Kevin P. Fleming wrote:
Kevin Walsh wrote:
The perpetual agreement grants the owner a non-cancellable right
to use changes and/or enhancements made to the Asterisk codebase as
[the] owner sees fit. As any Asterisk fork would, of course, be based
On Fri, Jul 22, 2005 at 05:39:43PM -0500, Carlos Chavez wrote:
I have an Asterisk server running todays CVS (updated it just in case
that was the problem). It has 3 X100P cards. The first two lines I use
as my normal work lines and the third is my fax line which I use with
SpanDSP. I
I try to track down an error that causes that Astcc just reports the time, but
not the costs.
I could narrow the problem down into this sub routine:
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
eval { my $adjtime = int(($answeredtime + $increment
[EMAIL PROTECTED] wrote:
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am
Asterisk wrote:
I've got several agents on a queue. However, they often forget to go
not ready or log off when they can't answer the phone.
I would like a person calling my queue to be on the queue for a max
of 2 minutes, and I'm using the rrmemory strategy.
I put a timeout of 12 on the
Is there any1 who has some experience with Asterisk in Turkey?
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On Fri, Jul 22, 2005 at 04:35:23PM -0700, Darren Parko wrote:
Hello,
I have an #include file containing user voicemail configurations.
This works fine for the most part, but when a user changes their
password via the phone the #include file is not updated.
Is there a way to do this?
I
Title: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware
It's Digium, not Dell.
I have two identical Dell 1850s, each with the allegedly offensive
built-in E100 Ethernet ports. I placed a TE410P card in each. One
worked great, the other would not
Dear Asterisk Community,
Asterisk 1.0 was released at Astricon 2004, in September last year. It's
been almost a year and we haven't been able to go ahead and release a
new version. Now is the time to try to move forward again.
As we've outlined before, the process is this:
The problem is in the line beginning with eval. It should read as follows.
my $adjtime = eval { $adjtime = int((($answeredtime -
$numdata-{includedseconds}) + $increment - 1) / $increment) * $increment;
return $adjtime };
Darren Wiebe
[EMAIL PROTECTED]
Ronald Wiplinger wrote:
I
On Sat, 2005-07-23 at 06:35 -0400, Joseph wrote:
exten = _6XXX,2,Busy
exten = _6XXX,3,Hangup
But the whole point is that I don't want the caller to hear a busy
signal or get hung up, I want the Queue to try the next available agent.
Which it does at the moment, just with the errors
The time stamps in ASTCC are useless as they are now:
Fri Jul 22 15:06:25 2005
Wouldn't it be better to use something like:
2005-07-22 15:06:24 Fri
I want to sort the records by date, but with the format now it is
impossible... or do I miss something?
bye
Ronald Wiplinger
Darren Wiebe wrote:
The problem is in the line beginning with eval. It should read as
follows.
my $adjtime = eval { $adjtime = int((($answeredtime -
$numdata-{includedseconds}) + $increment - 1) / $increment) *
$increment;
return $adjtime };
Thanks, I solved it by removing
On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote:
Adam Goryachev [EMAIL PROTECTED] wrote:
On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
For this reason, I believe that if a fork were
ever necessary, it would struggle to beat a distinct path away from
the Asterisk Binary
On Sat, 2005-07-23 at 12:00 +0300, Tzafrir Cohen wrote:
Disclaimers aside, who has the copyrights in those cases?
Do you actually read the emails on this list? or just like to jump right
in and help the brawl continue? The disclaimers don't affect copyright,
the author of the work/patch/source
You could give SJphone a shot too..
http://www.sjlabs.com/sjp.html
http://www.sjlabs.com/doc/SJphoneReadme.rtf -- the README for SJphone
On Fri, 2005-07-22 at 19:22, Time Bandit wrote:
Can anyone recommend a good soft phone that's easy to configure under
Asterisk and works well on a
The included seconds field is not taken into account when billing the
connect charge. IMHO this is a bug but I've not gotten enough feedback
to put the patch through. Therefore the patch has been closed. :-)
Darren Wiebe
[EMAIL PROTECTED]
Ronald Wiplinger wrote:
Darren Wiebe wrote:
The
Tzafrir Cohen [EMAIL PROTECTED] wrote:
Disclaimers aside, who has the copyrights in those cases?
Digium currently holds copyrights and/or is allowed to relicense the
full asterisk codebase as is currently distributed in the asterisk
tarballs on ftp.asterisk.org and also all the code in the
sorry to be a little off list,
but I am provisioning sipura spa-2100 with firmware
3.2.1
Finally they got it working with fullduplex
ethernet wan.
The wan ethernet mode can be configured only by
provisioning but I didn't find the provisioning field's name and the possible
values.
Is there
Hi Ceyhan,
We are specialized on Asterisk. Please check http://kulustur.com
Please also see
http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants
for other consultants in Turkey, especially EMEA and Europe sections.
I can try to answer your questions via email. But, people in this
On Sun, Jul 24, 2005 at 12:43:06AM +1000, Adam Goryachev wrote:
On Sat, 2005-07-23 at 12:00 +0300, Tzafrir Cohen wrote:
Disclaimers aside, who has the copyrights in those cases?
Do you actually read the emails on this list? or just like to jump right
in and help the brawl continue? The
On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote:
Adam Goryachev [EMAIL PROTECTED] wrote:
On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
For this reason, I believe that if a fork were
ever necessary, it would struggle to beat a distinct path away from
the Asterisk
We use SCSI on the Supermicro Dual Xeon and have no problems.
I think I will see what happens when I use the IDE CDROM.
Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Friday, July 22, 2005 9:45 PM
To: Asterisk
Hello all,
since we are sitated in Serbia and Montenegro and
Linux, VoIP,Open SourceandECONOMY hereare in the
diapers. We are traying to start "Asterisk implementation project" and
make it available to small people (in small companies,hotels,homes etc). The
reasonis that here we have
On Sat, 23 Jul 2005 13:21:56 +0300, Tzafrir Cohen wrote
On Fri, Jul 22, 2005 at 05:39:43PM -0500, Carlos Chavez wrote:
I have an Asterisk server running todays CVS (updated it just in case
that was the problem). It has 3 X100P cards. The first two lines I use
as my normal work lines
I'm not sure on sorting it but it is fairly easy to change in the source
code. Here is what you would have to do to change the format.
1. replace this line in astcc.agi:
$callstart = localtime();
with:
$callstart = timestamp();
2. Add this subroutine to astcc.agi:
sub timestamp() {
my
Olle,
Awesome! Now that everybody know your aiming for September 1 for
Asterisk 1.2, I'm sure will make it.
Come' on Asterisk community, step up to the plate!
PB
Olle E. Johansson wrote:
Dear Asterisk Community,
Asterisk 1.0 was released at Astricon 2004, in September last year. It's
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wiebe
Sent: Saturday, July 23, 2005 8:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ASTCC gives me only the time,
but no cost
The
Hello fellow asterisk people!
I have Asterisk listening on port 5061 and SER on port 5060.
Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
My problems are with SIP. I can make incoming calls from SIP to asterisk
and to any of the other networks, but when I try to make an
[EMAIL PROTECTED] wrote:
Hello fellow asterisk people!
I have Asterisk listening on port 5061 and SER on port 5060.
Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
My problems are with SIP. I can make incoming calls from SIP to asterisk
and to any of the other
I will have some extensions behind an E1. All of them will need the
features/applications of Asterisk.
Analog Extensions - PABX E1 - E1 Asterisk IP - VoIP trunk
^
|
|
M O wrote:
Hello,
Just chiming in here:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of
[EMAIL PROTECTED]
Sent: Viernes, 22 de Julio de 2005 01:24 p.m.
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Dell Hardware
Mmhh nice !! So,
I'm having trouble
with the latest cvs HEAD (7/22/05) and myWildcard TE405P I just got in
from Digium. I'm not able to get podprobe to work with the release. I get an
error "unable to install" however when I grab the stable it works great but no
realtime drivers for asterisk.
I also tried
The system has been rock solid on both performance and reliability. One is
located in out corporate office, two in local data center, and three
distributed California, Wyoming, and NYC. We have experienced issues with
some of our ISPs but nothing that could blamed on the hardware. I will
likely
Try Florz patch with your bristuffed asterisk. Better support for
missed interrupts.
Julian J. M.
On 7/22/05, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I tried to install a tdm400P and a monoBRI. I loaded zaphfc and wcfxs
modules and everything seemed allright but linux log shows the
Dear Helpers!
I have setup my asterisk with the iax phone. However, when I try to
dial out to the pstn phone from the iax phone, I can't hear the other phone ring
until they pickup the phone.
Doesn't anyone know the issue? Doesn't anyone has experience with the
problem? Please helping me or
On 23-Jul-05, at 11:22 AM, Kevin Walsh wrote:
On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote:
Adam Goryachev [EMAIL PROTECTED] wrote:
On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
For this reason, I believe that if a fork were
ever necessary, it would struggle to beat a
Dave Cotton wrote:
On Fri, 2005-07-22 at 15:42 -0500, Eric Wieling aka ManxPower wrote:
Eric Rees wrote:
We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX
sounds better then the land line.
Using
Thank you very, very much Rusty. I reopened the bug report.
http://bugs.digium.com/view.php?id=4479 I made a very slight change to
the method it uses to calculate costs but it should implement the
connect charge properly. Initially I rewrote the cost calculation code
but that was not
Kevin P. Fleming wrote:
Aidan Van Dyk wrote:
I guess I read 1 more FAQ than you. But I'm not sure how quoting a
Google FAQ is libelous FUD.
Your original message did not quote a Google FAQ, it accused Digium of
joining the SoC program only to collect the $500 mentoring fees. I
believe
Aidan Van Dyk wrote:
Is this indicative to how Digium people respond to everything (including the
company that built the first asterisk-supporting hardware still continuing
to make hardware which Asterisk works on)?
Nothwithstanding the almost-unparseable syntax, let's not feed this
troll
Hi Terry,
thanks for the info - might - so shall get an Intel board/cpu
combination or a quadbri card - or might it be the case, that the 2400+
athlon xp is running only as 1800+ cause I didn't update the mainboard bios?
yours,
Alex
Terry Wade wrote:
Spoke to Klaus-Peter about this PCI
Did you build it using the 64 bit CentOS or another Distro?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Dobrin
Sent: Friday, July 22, 2005 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello,
On Fri, 22 Jul 2005 19:49:15 -0400
Adam Dobrin [EMAIL PROTECTED] wrote:
| anyone else have the above issue? this is today's CVS.
|
| thanks.
Well, i found that as well couple days ago, and even reported that on
#asterisk-dev.
A full backtrace can be found here:
Sarge. RHEL/compatible should be fine too.
Wiley Siler wrote:
Did you build it using the 64 bit CentOS or another Distro?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Dobrin
Sent: Friday, July 22, 2005 4:47 PM
To: Asterisk
Aidan isn't a troll he does raise a very valid point.
/b
On Jul 23, 2005, at 5:55 PM, Brian Capouch wrote:
Aidan Van Dyk wrote:
Is this indicative to how Digium people respond to everything
(including the
company that built the first asterisk-supporting hardware still
continuing
to
Or better yet.. modify the disclaimer like I and a few others did to
say that the only thing you will disclaim are things you post on the
bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted
under your user on mantis IT IS NOT DISCLAIMED!
/b
On Jul 23, 2005, at 2:59 PM,
Brian West [EMAIL PROTECTED]:
Aidan isn't a troll he does raise a very valid point.
Which was, I presume, that companies that once collaborated on Asterisk
development such as Sangoma don't find themselves on friendly terms with
Digium now that they're competing for * implementors $$s?
If
Hello,
I just wanted to know if anybody is using callgen323 and ohphone.
I am running callgen323 on 2 machines with gatekeeper on one of the machines. When callgen is started, it registers with gatekeeper and places multiple call sessions. Ohphone has capability of placing single call and shows,
Title: Message
When using Postgres as the backend to voicemail.conf using
[general]
dboption=dbname=asteriskdb user=asterisk password=asterisk
Everything works great until I restart Postgres. In this case asterisk does
not attempt to reconnect to the database even when it knows that there is
Mine did.
[EMAIL PROTECTED] 7/21/2005 2:54 PM
Brian West wrote:
ClueCon is coming in 2 weeks so we urge everyone who plans on
attending to register today so we get a proper headcount!
snip
Thanks,
Brian West
Asterlink.com
snip
Anyone else think that was a joke at first impression?
Guys I just read on the wiki:
2005-07-19 - long awaited extension lights (hint priority) and call pickup
on various phones work with newly released asterisk patch digium bugtracker
- feel free to test and report findings to the bugtracker to have this
commited to cvs.
How does this work? And
option r. 'nuff said.
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+dial
On Sat, 23 Jul 2005 14:48:10 -0500, Maps [EMAIL PROTECTED] said:
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Also, both asterisks have notransfer?yes and I get this
-- Attempting native bridge of IAX2/[EMAIL PROTECTED] and
IAX2/voipjet-9
Why? Seems its not taking the notransfer into account.
Now Im puzzled
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf
Guys.
Which phones support pushing information or xml push info? For making info
applications and such, of course, besides Cisco?
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