Hello,
I'm using the classic [stdexten-macro] in extensions.conf whereby a call
is picked up by voicemail after a certain ringing time.
When programming a SIP phone to redirect calls (SIP 302 redirect) to
another extension I'd like to avoid that voicemail pickup so that the
call goes into the
Andrew Kohlsmith wrote:
Undue? Digium requires disclaimers so they can dual-license it for ABE and
other commercial vendors. You're purposely twisting and distorting the
reality with these weasel words.
I understand Digium strategy but I don't agree with it. I think it's
wrong not to
Kevin P. Fleming wrote:
The card manufacturer is irrelevant, as is the type of card. VMware does
not currently provide any sort of PCI bus passthrough to virtual machines..
Hopefully this will change soon.
--
Tomislav Parcina
[EMAIL PROTECTED]
___
Hi dear
is any web based sip-phone?opensource?
best
Mani
Food fight? Enjoy some healthy debate
in the Yahoo! Answers Food Drink QA.
http://answers.yahoo.com/dir/?link=listsid=396545367
Steve Underwood wrote:
I'll do it for 30% less than they quote. :-)
I didn't see on their pages, what is their price?
--
Tomislav Parcina
[EMAIL PROTECTED]
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SS == Stuart Sheldon [EMAIL PROTECTED] writes:
SS This might sound strange, but is there anyway for Asterisk to set
SS extra sip headers based on a sip phone returning a 302 in a
SS dialplan?
You can detect that a redirect has occurred by looking at ${RDNIS}.
You can't tell which SIP phone did
There is hotel application weary popular in Croatia - Micros-Fidelio.
Now I need to connect Asterisk with this application for purpose of
billing. Thing is that hotel would like to give customer one bill for
every service that he used while he was in hotel.
Has anybody connected Asterisk with
I think you can do this with outlook. Use the Third Lane dialer product,
set your extension to that of the conference, then initiate the calls.
It will call the extension then the party and connect the two.
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax:
Hello,
Is it possible to use ogg stream with asterisk as moh?
I have an icecast2 ogg streamer, but cannot use it with asterisk 1.4
The moh with files works
icecast2 works
but not icecast2+asterisk.
I think I need something like (see below) in music on hold config file:
mode=custom
[EMAIL PROTECTED] wrote:
I joined VirtualPhoneLine.Com service and am really enjoying the use of it.
I am pretty certain this constitutes fraudulent and *misrepresentative*
http://www.google.com/search?hl=ensa=Xoi=spellresnum=0ct=resultcd=1q=advertising+misrepresentativespell=1
advertising.
On 3/1/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tomislav Parèina wrote:
Is it possible to use Digium (or Sagnoma, or Beronet) cards with
Asterisk on Vmware?
The card manufacturer is irrelevant, as is the type of card. VMware does
not currently provide any sort of PCI bus passthrough to
Morten Isaksen ha scritto:
On 3/1/07, *Kevin P. Fleming* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Tomislav Parèina wrote:
Is it possible to use Digium (or Sagnoma, or Beronet) cards with
Asterisk on Vmware?
The card manufacturer is irrelevant, as is the type of
Use a Citel portico Telephone VoIP Adapter to interface the Meridian
phones direct to the Asterisk server http://www.citel.com/.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J French
Sent: 06 March 2007 00:04
To:
I am at the end of a long way... i try to work with a number of isdn
boards (BRI not PRI) and i found only a lot of problems.
First, the bristuff that is near working, but not so perfect ISDN
designed interface. This is not bad but in a production environment
this solution is not usable.
Second
The dialplan looks OK, depending of course on the numbers you trying to
dial. If you want the phone to wait for a given timeout period after the
digits are entered add a T immediately after the specific dialplan
rule. (ie: xx[2-9]xxT). I'm assuming from your rules you need to
dial a 9
Tomislav Parcina wrote:
Thomas Kenyon wrote:
Asterisk also seems to barf if it makes a registration/renewal request
and it doesn't receive a reply in a timely fashion which will
obviously happen if your internet connection disappears. (all versions
I've used).
That's why people should use
»Steven Ringwald« wrote:
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written
for this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of
Axel Thimm wrote:
Get it from here: http://atrpms.net/dist/el4/speex/, or since your
using a yum based distribution, point yum to atrpms and let it do the
work.
They don't have 1.2.x version there?
How fast do they make package since source version is out?
--
Tomislav Parcina
[EMAIL
Tomislav Parcina wrote:
They don't have 1.2.x version there?
Newer mind, I found it :)
How fast do they make package since source version is out?
This question still stands.
--
Tomislav Parcina
[EMAIL PROTECTED]
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DST rules can be found by searching the sip.cfgg file for SNTP.
You will find a cluster of time parameters, including the month and
day upon which to change DST.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
iaxtel:
Am Dienstag, den 06.03.2007, 05:18 -0400 schrieb Chris Mason (Lists):
Of course, it would be highly unlikely anyone on the list would want
to report Rehan...but in case anyone does:
I have been told that unsolicited commercial e-mail (I do not imply that
Rehan's post fulfills the criteria,
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get auto-answered. However, I have
Hey,
Implementing Asterisk on local Lan spread over 2 campuses on two different
cities is our graduation project.
Having done all the research and reading stuff. I started with the practical
work.
Not getting a hand on the linux digium. I installed Red Hat linux 9.0. I was
able
On Tue, 6 Mar 2007 12:22:24 + (GMT)
Asterisk Asterisk [EMAIL PROTECTED] wrote:
Hey,
Implementing Asterisk on local Lan spread over 2 campuses on two different
cities is our graduation project.
Having done all the research and reading stuff. I started with the
practical
On Tue, Mar 06, 2007 at 11:59:10AM +0100, Tomislav Parcina wrote:
Tomislav Parcina wrote:
They don't have 1.2.x version there?
Newer mind, I found it :)
How fast do they make package since source version is out?
This question still stands.
As fast as they read asterisk-announce ;)
--
Hello,
I am having something of an odd problem: about every 100 calls or so,
when a call comes in via an external mISDN interface and I route it to
an internal mISDN interface by dialing an internal msn that is
programmed for multiple phones on the internal bus, somtimes the other
phones
Louis-David Mitterrand wrote:
Hello,
I'm using the classic [stdexten-macro] in extensions.conf whereby a call
is picked up by voicemail after a certain ringing time.
When programming a SIP phone to redirect calls (SIP 302 redirect) to
another extension I'd like to avoid that voicemail
Hi,guys. I try to install zaptel 1.4 on dell 2850, my OS is fc6, I can compile
and install it with no error. But when I modprobe wd4xxp module, the OS hang.
pI must push the reset button on 2850's panel to reboot the OS. but when OS
starting udev, it can't continue, that is, I can't enter the
Here is some of my actual Polycom config files. The only thing that has
been changed is the hostname of the PBX. We assign the SIP user ID as
the MAC of the phone with a -a -b -c, etc appended to it for each of the
line appearances. These config files do not have the DST stuff added yet.
hello,
i have configured internal pickup my problem is when an external call is
coming i cannot pickup coz the extension is an external number.
isit possible to pickup the external via n+101 prio or is ther any other
solution?
my config:
exten = _*8.,1,GoToIf($[${CDR(userfield)} =
James FitzGibbon wrote:
They say it's custom made for them, and I certainly can't find anything
else like it after several hours of searching, but it seems to be what's
required. I'll have to rig up a backplate with a cutout to get the 12V
connector into the case, but other than that I'm
Juraj Bednar wrote:
I'm looking for a way to have G.729 codec working on Solaris/x86.
Binary codec from Digium is not compiled for Solaris/x86 (only sparc).
Are there any alternative (free or commercial) G.729 implementations,
which would work?
We will have Solaris 10 x86-32 and x86-64
Mr. James W. Laferriere wrote:
Hello All , I'd usually just take the latest timestamped tarballs
use them , But this has gotten me a tad setback .
I want to build astersik-1.4.1 I am not sure which of these is
going to work correctly . Anyone else have a better idea than me ?
You
On Tue, Mar 06, 2007 at 07:18:08AM -0600, Eric ManxPower Wieling wrote:
How can I detect that a call has been redirected and should no longer be
intercepted by vm?
That should happen by default. The call should get sent to the new
place and it should act like the call was directly
On Mar 6, 2007, at 1:55 AM, Tomislav Parcina wrote:
Kristian Kielhofner wrote:
Hey everyone,
I came across a situation where I needed to use CDP to advertise a
voice vlan to Polycom/Cisco (and other CDP capable phones) without a
Cisco switch.
Hi Kristian!
Thank you for your work. I'm not
I'm running chan_cellphone version 13 on the latest svn trunk (as
root). I believe I have chan_cellphone set up correctly (bt addr and
port retrieved from the cell search CLI command). When I load the
chan_cellphone module, my Motorola V3m asks if I want to allow Asterisk
PBX, I say yes and
On Tue, Mar 06, 2007 at 09:22:44PM +0800, [EMAIL PROTECTED] wrote:
Hi,guys. I try to install zaptel 1.4 on dell 2850, my OS is fc6, I can
compile and install it with no error. But when I modprobe wd4xxp
wc4xxp ?
module, the OS hang. I must push the reset button on 2850's panel to
reboot
Hi!
Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to
show the Caller number on the phone.
There's a Caller ID Method: option on Regional settings, but I
tested all options, and my CLIP phone never shows the Caller number...
:(
Any idea?
»Steven Ringwald« wrote:
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written for
this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of
Any ideas as to how I can fix this issue?
Thanks
Remi
Remi Quezada wrote:
Ok that makes sense, but I'm still getting double digits. It seems to
me that the DTMF digit is getting detected too late. When the digit is
pressed it seems like asterisk is passing the DTMF digit for a fraction
of
Hi, all,
I am using Asterisk 1.2.15 with an OpenLine4 card (vpb-driver 4.0). And
Asterisk segfaults. Here is the
output of loading chan_vpb. Very detailed because I turned on vpb
verbose. any lead to solution will be
appreciated. Thanks
output from Asterisk:
[chan_vpb.so] = (VoiceTronix
Hello
This is another method, if you don want to change zaptel.conf (from
Chris Hozian of Digium)
I would like to clarify that there is another method which may be used
that does not require you to load the kernel modules in a different
order or to modify your zaptel.conf and zapata.conf file.
In 1.2, try adding noload = res_musiconhold.so to your modules.conf.
In 1.4 though, it would be worth a try, but I don't know for sure if
that's how it's done.
Moj
David Thomas wrote:
On 3/5/07, C F [EMAIL PROTECTED] wrote:
Could be its trying but does it actualy play the music?
It's not
i've submittet the project to SF.
I have to wait 2 business days for their validation.
The project is in a beta release and will be released on GPL.
Bye
On 3/2/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
nik600 wrote:
i'm sorry but due to some problem the software will be released not
Bill Gibbs wrote:
Sorry to reply to myself, once again onn the list, but since SLA is new
I figured I should answer my own question before anyone else gets
confused...I completely forgot about my -directory.xml defaults...so
that's where all these bogus SUBSCRIBE requests were coming from.
Doug Lytle wrote:
»Steven Ringwald« wrote:
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written
for this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not
to
Bill Gibbs wrote:
I have been using 2 Polycom 430s so far. I can get incoming calls just
fine (both phones ring on line 1). However it doesn't appear to seize
the line, so if a call is on the one phone, I can still pick up line 1
on the other and dial - and it's reflected in the connected
On 3/6/07, Russell Bryant [EMAIL PROTECTED] wrote:
This will connect the
station to the first available trunk if there is one, and then provide
dialtone for making a call.
That's what I was concerned about. Whether it connects to the first
available, or the first one. In other words, if
Friends in the Asterisk community,
One thing I avoided working with for a long time is the Asterisk
voicemail code. One module in
Asterisk I've constantly been naming as one of the worst parts is
voicemail. One part of
Asterisk that I've been kind of avoiding during my trainings is
I think it has something to do with hints...I can't seem to subscribe to
anything now with 1.4 vs 1.2, even with a normal non SLA setup.
My phone/config that works with 1.2, so I know hints work with the phone
and firmware and with NAT at least on 1.2.
I did a fresh 1.4 install (and I did a make
I've used the Dock 'n Talk, and I can say that it worked as well for us
as it claimed to be able to. Only need an analog Zaptel card of some
sort. I know there are a few other brands available, as well as some
GSM Bridges available that you insert the SIM card directly into,
bypassing the
I assume my SUBSCRIBE issue for hints has something to do with this bug
http://bugs.digium.com/view.php?id=9168
Bill
snipped previous emails for readability
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Anyone know the gsm encoding mip requirement from g711? Or number of
channels can be transcoded from g711 to gsm at a time.
Thnx
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Every few seconds I get the following message:
== Parsing '/etc/asterisk/manager.conf': Found
== Connect attempt from '127.0.0.1' unable to authenticate
I'm trying to track down where it's coming from.
I've used TCPDUMP NGREP to monitor 127.0.0.1, no data's flowing.
I've tried
Hi Ken,
Trixbox comes with the Flash Operator Panel. The FOP server is likely setup
with incorrect authentication parameters, and hence is failing
authentication everytime it attempts to use the Asterisk Manager API to
update it's tracking of what's going on in your system.
Check your
We installed a quad xeon 3ghz which transcoded ~100 active channels (as a
gateway). Take a look at the codec demands (in asterisk show codecs I
believe) and scale from there. This box was 60% loaded - which is all we're
comfortable with before latency goes too high.
Michelle Dupuis
Technical
That was indeed the problem. I thought I had eliminated any httpd by
disabling the service, but the problem was trixbox was still trying to
load on startup via /usr/sbin/amportal. Once I removed that from
startup, problem resolved.
I did go a step further and wipe out the panel in httpd as
On Tue, Mar 06, 2007 at 02:26:21PM -0700, Ken Williams wrote:
That was indeed the problem. I thought I had eliminated any httpd by
disabling the service, but the problem was trixbox was still trying to
load on startup via /usr/sbin/amportal. Once I removed that from
startup, problem
How to compile smsq in 1.2? It is compile in 1.4 by default. It is
included in 1.2.13, but not compiled. Any rule or method to make it?
Yuan Liu
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To
Hello,
I'm wondering how one might set up a feature to add (in real-time) the
current CallerID information to a junk-callers database.
After answering a call from an outside line and determining that the
call was from a telemarketer or the like, the user could dial an easy
specific code
Hello,
I'm wondering how to make it possible for the user to cancel the last
entered digit, if he made a mistake.
For example, a user calls and starts entering 1...2...4, then he should
be able to press, lets say *, to cancel 4 and enter i.e. 3.
Thanks
Jake
--
--- Domeny w ULTRA
Why not just push * to start again. The customer most of the time wont
know that they made a mistake until you are reading the digits back to
them anyway.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED]
Will try that again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Parcina
Sent: Tuesday, March 06, 2007 11:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: build rpm fails
Axel Thimm wrote:
Get it from here:
Lacy Moore - Aspendora wrote:
That's what I was concerned about. Whether it connects to the first
available, or the first one. In other words, if line 1 is in use,
does it connect to line 1 or line 2?
If you take a phone off hook without pressing a line button, and the
phone is properly
Dave Fullerton wrote on 3/6/07 9:33 AM:
Polycom's 2.1.0 firmware has the new DST settings as the default. This
is what they use for the SNTP element:
SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=
tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=
uname -a
Linux xxx 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:37:32 EDT 2006 i686 i686
i386 GNU/Linux
modinfo zaptel | grep ^version:
version:1.4.0
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I can run zaptel 1.4 normally in other machine on the same OS, only can't run
it on 2850. It hangs the OS.___
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exten = 111,1,Wait(1)
exten = 111,2,Playback(Randy)
exten = 111,3,Dial(Sip/Randy,20)
exten = 111,4,Goto(111-${DIALSTATUS},1)
exten = 111-BUSY,1,Voicemail([EMAIL PROTECTED],u)
exten = 111-NOANSWER,1,Dial(IAX2/${TELIAX_OUT}/212551212)
works GREAT
This is awesome, I had actually wondered about
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller
Sent: Tuesday, March 06, 2007 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Questions
Dave Fullerton wrote on 3/6/07 9:33 AM:
I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got
two-way audio between Google Talk and Asterisk! This IS an exciting moment
today in VoIP! This is just GREAT!
- Ronald Lewis
http://ronaldlewis.com
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Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I
am totally floored with how cool it is!
Thanks,
Steve Totaro
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I guess I would not consider this an advanced application. I have
something that will do this will all conversations, not just voicemails.
Voicemail should be trivial.
Thanks,
Steve Totaro
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent:
J French wrote:
We have a dinosaur Meridian system (~version 2) with 4 digital lines
going to a Repartee Voicemail server. The Repartee got smoked by
lightning two days ago and I'm itching to get Asterisk installed in
its place. PRI is not an option since the system is so old that it
doesn't
After upgrading to Asterisk 1.4.1 from 1.4.0 it just worked for me.
There must have been a bug in 1.4.0. I have successfully connected to a
Gmail and MSN instant message client.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mani
Sridhar
Sent: Saturday,
International calls (Germany) haven't completed since around 3/1. Domestic
works. Is it just me? I'm getting 503 responses.
Tom
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Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to
show the Caller number on the phone.
There's a Caller ID Method: option on Regional settings, but I
tested all options, and my CLIP phone never shows the Caller number...
It should work fine.
First, verify that you have for
--
Message: 1
Date: Tue, 6 Mar 2007 20:02:07 +0100
From: Olle E Johansson [EMAIL PROTECTED]
Subject: [asterisk-users] Building a new voicemail system... Testers
needed!
To: Asterisk Non-Commercial Discussion Users Mailing
Hi Team,
I have integrated asterisk with Toshiba analog PBX. NOw the live setup is
going.
Now I am facing call droping problem. It's happening ample time. 10-20 calls
are droping every day.
What could be the reason. I attached latest zapata.conf file for your
information.
This is being a
Everything can be done with a certain amount of coding... :-) No,
it's not possible
in Asterisk today.
Check the configuration templates to make life easier when
configuring voicemail.
It's documented in doc/configuration.txt in your asterisk source code
directory, or
here:
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