Hagen Rode wrote:
Hi
I'm browsing around the internet looking for signs that the IAX client
library and app_meetme support video.
I stumbled across this post by SteveK on the 27th of Feb 2006.
My company is looking to hire a full-time developer, who will be working
about 25-50% of
On Thu, Mar 02, 2006 at 02:32:43PM -0600, Anton Krall wrote:
|Try this:
| http://www.bicomsystems.com/docs/pbxware/
Looks very nice.. Is it GPL, GNU?
Nither the GPL nor any other free software license.
google for pbxware.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
On Thursday 02 March 2006 22:19, Matt Schulte wrote:
All, I'm not sure how to word this question but we're noticing a lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used
to seeing 8+ .. There is no
Martin Joseph wrote:
On Mar 2, 2006, at 3:46 PM, Wai Wu wrote:
You can really mix G729 encoded frames. So I would guess that licenses
are not needed for non-G279 devices. BTW, there is a difference
conference app (forgot the name) that only mixes the two parties that
have the loudest
amaury BOSSE wrote:
Is there a free linux tool which can test voip call quality between two
Asterisk PBX.
It will help me to test the WAN network between them.
I have only found commercials ones, so if you know a free one, let me
know.
For packet loss, rtt etc and a phone call check out:
On 4 Mar 2006, at 08:30, Paul Hewlett wrote:
On Thursday 02 March 2006 22:19, Matt Schulte wrote:
All, I'm not sure how to word this question but we're noticing a
lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID
Hello everyone.
My land-line provider (Romtelecom) has a very nice offer for ISDN. All in
all they offer me a digital land-line with 1 base number + 2 MSN's and that
would make a grate addition to my full-time home office.
Romtelecom say they're providing EURO-ISDN and the line is compatible
Hi. I am using 1.2.5 and I have an extension using a zap fxs channel
on a 400P Digium card. Now when thatextension is dialed with a
timeout of 20 seconds it rings for about half a second and then the
log says noone picked on after 2 seconds and so it goes to
voicemail.
Any assistance would
Hi everyone,
Could any tell me How i can accept unregistered
Gatekeepers calls to my Asterisk Box?
My customer is using another Gatekeeper and he want to
use my Asterisk as a gateway for him to terminate the
call using SIP protocol. and his Gatekeeper is not
supported as end point to register my
HI There
if the line is a standard EUROISDN with DSS1 protocol then it's not a risk
at all to take and to connect it. You get an S0 interface from your TELCO
and there cou can plug in any EUROISDN compliant equipment e.g. TAs, phones
and, of course, ASTERISK ...
I would suggest that you AT
I have been working with authenticated notifys for auto resync my
autoprovisined devices.
But it seems to stop the state machine, and when Endpoint answers 401
Unauthorized, the Sip Notify command from cli, does not answer with a
Authenticated Notify?
Have i misconfigured something?
The line is supposed to be standard EUROISDN; I found no mention of DSS1 on
the technical specs page, I only found EUROISDN.
The TELCO is going to provide me with a NT equipment that has two analog
ports and two S0 ports; Of the two S0 ports one is supposed to be used to
connect the PBX to the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Bill Gibbs wrote:
All I have found was stuff about softkey templates in Call Manager. If
there is any programming we could do without CM that would be
fantastic!! For some reason I can?t get an iDivert key to show up on my
7940G!
The SIP
On 03:10, Sat 04 Mar 06, Tzafrir Cohen wrote:
On Tue, Feb 28, 2006 at 05:25:40PM -0700, Damon Estep wrote:
Try nat=yes and qualify=yes in sip.conf.
So a call between two SIP phones will have to go through the remote
server? Or can those two phones be aware of each other?
Yes. But without
On 10:34, Sat 04 Mar 06, Ron Wellsted wrote:
The SIP firmware does not allow the softkeys to be programmed :(
Unfortunately you have to make a choice:
SIP firmware - Easy to implement on *, but poor XML support
SCCP firmware - poor/non-trivial asterisk support, great XML support.
I have to
The line is supposed to be standard EUROISDN; I found no mention of DSS1 on
the technical specs page, I only found EUROISDN.
OK, it should be DSS1 then ...
The TELCO is going to provide me with a NT equipment that has two analog
ports and two S0 ports; Of the two S0 ports one is supposed to
I have a similar configuration: two Alcatel PCX 4400
with E1+DID anda dialplan shared between sites.
How do you plan to configure Asterisk boxes to share
dialplan?
DUNDI?
Thanks for any info
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hafez
AzzamSent:
Awesome.
Any URLs to the XML templates for all the features?
The SCCP firmware doesn't appear to have any in the zip file.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent: Saturday, March 04, 2006 6:07 AM
To:
HELLO everyone
I am having two alcatel 4600 digital phone PBXs .. They are situated in
two
locations 15km apart.
I want users or extension in both PBXs to be able to dial and receive
calls
from each others through those 30 channels in the E1 ..
I have line of sight so i am planing to
Dear All,
I am new to both linux and asterisk. i want to install asterisk on suse 10 but
receiving the error: termcap support not found. i dont know what to do?
i shall be highly obliged if someone helps me.
zeeshan
___
--Bandwidth and Colocation
On 07:39, Sat 04 Mar 06, Bill Gibbs wrote:
Awesome.
Any URLs to the XML templates for all the features?
The SCCP firmware doesn't appear to have any in the zip file.
Have a look here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services
good luck
--
Michiel van Baak
[EMAIL
Duh. Thanks. I spent all my time looking for SIP XML configs that my
brain is fried now.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent: Saturday, March 04, 2006 8:14 AM
To: asterisk-users@lists.digium.com
Subject: Re:
On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote:
I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811..
I'm running 1.2.1 and most of mine get cut short too.
On Thu, Mar 02, 2006 at 02:19:29PM -0600, Matt Schulte wrote:
All, I'm not sure how to word this question but we're noticing a lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used
to seeing 8+ ..
Have you tried dialing an 800 number? Does that work? This extension:
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
seems to be missing one X since it's only 10 digits long. Your PSTN
probably requires a 1 to be dialed also. On the other hand,
exten =
In Sweden, where I live, it's snowing like crazy. The Stockholm area
is covered in white stuff
and there's really no reason to leave the computer and get out
anywhere. More white stuff
is coming down all the time. Boring. I am sure your weekend is no
better - rain, snow or
just another
Greetings,
I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be
Thanks for the tip!
I shoud have found this on my own...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Friday, March 03, 2006 5:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
My dial plan is as simple as it gets:
exten = 101,1,Dial(sip/sip101,180,Ttr)
But I'm doing blind transfers and you're doing attended transfers.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dinesh Nair
Sent: Saturday, March 04,
Asterisk 1.2.5 is now available for download on the ftp. See the
ChangeLog for details about what has changed.
ftp://ftp.digium.com/pub/telephony/asterisk/
As mentioned in the release announcement for Zaptel 1.2.4, our releases
now contain some extra files. The Asterisk release is available as
HiDoes someone have a better sql query for selecting the provider used by LCDial application than the one proposed in the tgz ? It's far from working well with most of price lists.I tried to tweak it somehow with more or less success.Regards, Michel -- Michel Luczak[EMAIL
I actually got it all working - but it's great to see where we did the
same
thing, and where we differ.
I ended up using the 'pop' perl command - inside a loop to go back one
item
at a time through my list
PaulH
Nice work! Perl = TMTOWTDI = There's More Than One Way To Do It
-MC
Message: 6
Date: Fri, 03 Mar 2006 17:32:47 +
From: Conrad Wood
[EMAIL PROTECTED]@[EMAIL PROTECTED]@conradwood.net
Subject: Re: [Asterisk-Users] Problem with NAT!!!
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
The problem is the remote server. Asterisk is able to drop the media stream and
allow the SIP phones to communicate directly, which has both its drawbacks and
advantages depending on how you plan to use asterisk. For this to take place
you'll need the planets to be in the proper alignment and
Someone urged us to implement this behavior. I guess there was a large
company that told us that they were not able to send another MWI that
indicates that the messages were deleted... So far people could live
with this smart idea (it was not our idea).
CS (yes I am from snom)
-Original
On Sat, 2006-03-04 at 13:04 +, (pg) Zeeshan wrote:
Dear All,
I am new to both linux and asterisk. i want to install asterisk on suse 10
but receiving the error: termcap support not found. i dont know what to do?
i shall be highly obliged if someone helps me.
zeeshan
Are you
The only two programmable buttons are the 'Messages' and 'Services'
and 'Directory buttons'. They are all configured in sip_default:
messages_uri: number to dial
services_url: xml file to load
directory_url: xml file to load
Cheers,
Omar
On 3/4/06, Kevin Steil [EMAIL PROTECTED] wrote:
Does
On Mar 4, 2006, at 7:54 AM, The Asterisk Development Team wrote:
Asterisk 1.2.5 is now available for download on the ftp. See the
ChangeLog for details about what has changed.
Reading the changelog I notice the following... I suppose it should
say incorrect?
2006-02-17 01:55 +
Hello,
You should install (with YAST2) termcap. And also mpg123, which is not
included in the distro... Otherwise, you can simply install asterisk from
Yast2 directly (but older version: 1.0.9.4).
Alban
Le Samedi 4 Mars 2006 18:55, Pete Barnwell a écrit :
On Sat, 2006-03-04 at 13:04 +,
Someone urged us to implement this behavior. I guess there was a large
company that told us that they were not able to send another MWI that
indicates that the messages were deleted... So far people could live
with this smart idea (it was not our idea).
I don't understand why you have to be
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9XX,2,Congestion()
exten = _9XX,102,Congestion()
I think these 3 lines need to have a 1 added like this:
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/1${EXTEN:1})
exten = _9XX,2,Congestion()
exten =
At 10:03 PM 03/03/2006, you wrote:
You mean like this
exten = ww_9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
thanks
More likely:
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})
Ira
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 /
We have a bunch of PAP2s, and using the # to send immediately does not
work as described in the manual. The PAP still waits for the
Interdigit_Short_Timer to expire before sending the dial string. In
addition, the dialplan does not cause the string to be sent
immediately as it should.
Here's
I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of
the site. It is sending CRC errors )to Telsta, drops all calls once a day
for 1 second, calls getting stuck, quite unpleasant!
I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri.
James
-Original
Probably just me being dumb, but I am trying to update my asterisk to
the latest (1.2.5)
When I go to my /usr/src/asterisk I type:
make upgrade
make install
This seems to be doing it's thing, but when I type show version from
the console (after a restart) I still get:
Asterisk
Martin Joseph wrote:
Probably just me being dumb, but I am trying to update my asterisk to
the latest (1.2.5)
When I go to my /usr/src/asterisk I type:
make upgrade
make install
This seems to be doing it's thing, but when I type show version from the
console (after a restart) I still
Hi all,
I hope everyone is doing well. I just joined the list, and I've really
enjoyed all I have read about asterisk so far. Unfortunately, I'm having a
bit of trouble implementing this thing :).
By the way ... I did my best to search the forums, and also to use google
extensively, and while I
I think you need to:
- pull the 1.2.5 tar.gz file from ftp/http
- extract it into a dir (tar xvfz filename)
- cd into it
- excecute make upgrade and make install
Cheers,
John
[EMAIL PROTECTED] 04-03-06 21:12
Probably just me being dumb, but I am trying to update my asterisk to
the latest
On Mar 3, 2006, at 1:46 PM, Gavin Adams wrote:
Hi All,
I'm stumped on a weird problem. I have an * server working fine for
local
SIP phones and IAX2 connections. We just provisioned a second Ethernet
port to attach to a local SIP provider.
PSTN calls incoming work fine:
PSTN - SIP
Bill Gibbs wrote:
Vim for everything
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pete
Barnwell
Sent: Friday, March 03, 2006 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Preferred editor(s)
Downgrade to 1.0.10. I was unable to get the 12sp+ to work reliably in 1.2.0-1.2.4 and had the same problem.On 2/20/06, btb
[EMAIL PROTECTED] wrote:hello-i'm having trouble completing a connection between an older skinny
phone (12sp+) and a soft sip phone (x-lite).the skinny phone appears to
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote:
Unfortunately you have to make a choice:
SIP firmware - Easy to implement on *, but poor XML support
SCCP firmware - poor/non-trivial asterisk support, great XML support.
The newest SIP firmware (beta versions) allows the exact XML
Hey everyone,
We have a special mail box for certain customers when we are out of the
office. Basically they enter a pin number and if it is valid they leave
a message and it notifies the on call techs. My question is regarding
externnotify for the voice mail.conf. If I enabled that and set
Thanks. I will try. Is there any documentation thatdescribes this fix? I cant find it anywhere in any docs.Joseph Tanner [EMAIL PROTECTED] wrote: Like this:exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})Joseph TannerOn 3/4/06, sdgesa gaeharth <[EMAIL PROTECTED]>wrote: You mean like
In our area code(703), and I am not sure if it is like this in other places, we are required to dial the area code even if we dial local numbers . That is what these lines are for:exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})exten = _9XX,2,Congestion()exten =
Funnily - I have set up 2 or 3 pri's over the last few weeks on 1.2x and
haven't had any issues.
(and one of those is a high load situation - passthru at an outbound call
centre)
PaulH
Melbourne
- Original Message -
From: James Sturges [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List
vi here
On 3/4/06, JP Carballo [EMAIL PROTECTED] wrote:
Bill Gibbs wrote:
Vim for everything
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pete
Barnwell
Sent: Friday, March 03, 2006 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Testing Asterisk doenst make for a boring day at all :)
On 3/4/06, Olle E Johansson [EMAIL PROTECTED] wrote:
In Sweden, where I live, it's snowing like crazy. The Stockholm area
is covered in white stuff
and there's really no reason to leave the computer and get out
anywhere. More white stuff
More and more areas of the US require 10 digit local dialing, and 11
digit toll dialing.
Unfortunately, that isn't universally true. Some states have decreed
that 11 digits will be dialed for local and toll, other locales have 7
digit dialing across state lines, and at least one location,
You should also know that this ONLY works with DTMF on analog lines. If
one happens to have to use pulse dial on a POTS line, there is no way to
delay dialing, and Asterisk STILL will not wait for dialtone, since no
one who is able to fix it seems interested.
John Novack
sdgesa gaeharth
a message and it notifies the on call techs. My question is regarding
externnotify for the voice mail.conf. If I enabled that and set up a
call file, will it do it for every voice mail box I have on the system?
Is there a way I can limit it to just the one voice mail box on the
system? If
Sounds great..thanks...
-Original Message-
From: Greg Oliver [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 04, 2006 8:25 PM
To: [EMAIL PROTECTED]; Asterisk User List
Subject: Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted
On Sat, 4 Mar 2006, Greg Oliver wrote:
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote:
Unfortunately you have to make a choice:
SIP firmware - Easy to implement on *, but poor XML support
SCCP firmware - poor/non-trivial asterisk support, great XML support.
The newest SIP firmware (beta
On 03/04/06 16:30 Paul Hewlett said the following:
On 2.4 kernels you would be using the LinuxThreads implementation of POSIX
threads. This emulated the POSIX threading model with some limitations -
to continue with this thread (pun intended !) and for freebsd users, the
default asterisk
On 03/04/06 23:54 The Asterisk Development Team said the following:
However, there is also a patch against the
previous release as an option for a smaller download,
asterisk-1.2.5-patch.gz.
well done, this makes it a lot easier on the downloads for those closely
tracking the releases.
--
On 03/04/06 23:17 Cosmin Prund said the following:
My dial plan is as simple as it gets:
exten = 101,1,Dial(sip/sip101,180,Ttr)
But I'm doing blind transfers and you're doing attended transfers.
oh right, i had misadverntly thought you were doing attended xfers as well.
with blind xfers,
Douglas Garstang wrote:
I have ztdummy installed:
Module Size Used by
ztdummy 3464 0
zaptel218756 1 ztdummy
crc_ccitt 2176 1 zaptel
ohci_hcd 16388 0
floppy 49028 0
pcspkr
68 matches
Mail list logo
