Ronald Wiplinger wrote:
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots of monkeys
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Deadlocks are not a config or Trixbox issue.
I'm confirming this one!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
Don't forget to analyze the QUEUESTATUS variable. :)
--
i've just set
joinempty=no
and then in extension.conf :
exten = 2701,1,Queue(2701|t|||10)
exten = 2701,2,Background(orario_2701)
Thanks to all
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On 11/14/06, Vicky [EMAIL PROTECTED] wrote:
One more thing i would like to point out is that softphones like sjphone use
some freeware stun server to detect nat on network (as a client ) . Asterisk
( as client ) cannot use external stun server to detect nat type
automatically so i think thats
Hi,
Has anyone tested Siemens Gigaset SL75 with Asterisk ?
How would you rate its performances ?
Cheers
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Michiel van Baak a écrit :
Hi,
Hello
Anyone here has any experience with the Nokia E70 and
asterisk ?
I read on the nokia website this phone is capable of talking
SIP and do Presence based on SIP/SIMPLE.
Please share your experience, I'm thinking of getting one
but want to be sure I can
Hi,
I have an application where we need to route several channels of a PRI
going into the Digium TE110P card on one Asterisk server to another Asterisk
server. I am thinking that this might be possible by using TDMoE and somehow
bridge the corresponding channels from the Zap channels of the
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I want to guess that it's your SIP provider.
Faxing via VoIP (SIP) is not reliable unless you are using T38, my guess
is your sip provider is providing this feature.
Hi Doug!
Have I understand it right. You are saying that my
Eric ManxPower Wieling[EMAIL PROTECTED] Wrote on: 11/16/2006 7:36 PM:
Special extensions like a, o, i, etc do not seem to be read from
include = 'ed contexts.
Is this a bug or as designed?
In this case, it is not an include. The i seems to work fine. From an off
list discussion, it
Hope this is the right place to report/ask for help...
Have have a 1.2.7.1 installation running reasonably happily for a while.
Thought we might give 1.4.0b3 a go. Ran it on a local test machine (that
has the single port card) and all was well. However, when I run it on a
machine with the
A little progress on this problem
Examining the logs i found a weird looking 'soft hangup' which
reminded me on an earlier issue we had. (and the reason why we were
still on the 'i' release of bristuff). It looked as if the channel
hung up just before rxfax actually could begin to work.
I am testing asterisk (version 1.2.12.1) on a Dell 1950 server and have this
strange problem on music on hold.
When I called into a queue using SIP from PSTN line which goes through our
cisco gateway (cisco 5300), asterisk will start play music on hold. But this
MOH seems at voice activation
Hi,
Do you have vad disabled in your dial-peer voice XX voip dial-peer?
What kind of MOH are you using; asterisk native or an external player like
mpg123?
--basv
On Fri, Nov 17, 2006 at 08:41:49AM -0500, gc wrote:
I am testing asterisk (version 1.2.12.1) on a Dell 1950 server and have this
Andrew Joakimsen wrote:
Has anyone gotten Asterisk to compile on Solaris 10? I have tried both
1.2 and 1.4 and I get errors about editline. Actually it seems that
1.4 goes through more of the process, but thats not good enough
Hi
i am planning to install Asterisk with a WildCard TDM400P on a redhat
enterprise 3.
I will use the last stable source, 1.2.13 Asterisk and 1.2.11 Zaptel.
Do you know if there are some issue with this version?
Can i compile Asterisk with the classic make make install make sample?
Do you
Would you mind explaining how you got it to compile?
Regards,
Andrew
On 11/17/06, J. Oquendo [EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
Has anyone gotten Asterisk to compile on Solaris 10? I have tried both
1.2 and 1.4 and I get errors about editline. Actually it seems that
1.4 goes
Andrew Joakimsen wrote:
Would you mind explaining how you got it to compile?
Regards,
Andrew
I didn't do anything out of the ordinary. I followed
https://svn.sunlabs.com/svn/solaris-asterisk/README to the letter...
PATH=/usr/sbin:/usr/bin:/usr/ccs/bin:/usr/sfw/bin
svn co
Andrew Joakimsen wrote:
Would you mind explaining how you got it to compile?
Regards,
Andrew
http://www.infiltrated.net/asteriskSol.tar.gz if it makes it easier for you
-bash2-2.05b$ ls -ltha asteriskSol.tar.gz |md5
38e6eb68e2ef29968a61b16fca2e3c78
--
Andrew Joakimsen wrote:
Would you mind explaining how you got it to compile?
Regards,
Andrew
See what happens when you're overdosing on coffee.. Anyhow:
SunOS *sun4u sparc *SUNW,Sun-Fire-280R
Take note for others downloading... It's not x86 Solaris before someone
shoots off It won't
El vie, nov 17 de 2006 a las 04:34 +0530, Vicky comentaba:
Thats really strange .. if you have made canreinvite=no then it should not
even attampt native bridging and should transcode codecs ..something's fishy
here .. Also try to put canreinvite=no in testulaw exntension too .
So why do I
Hi List,
I am looking for a 1 FXO analog termination device, other than the
obvious PC + FXO card, and which can achieve decent call quality. The
SPA-3000 seems an option... have you got any other ideas?
Cheers,
Jean-Michel.
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Hello,
I have a few users using Polycom 501's and some are using the forward
function on the phone. When a call comes in the system, he/she gets the
standard welcome to abc inc., bla bla bla message. When the dial an
extension, they get forward to the phone, which forwards them to a cell
Hi,
My pstn provider is currently set up so that when asterisk sends an outbound
SIP call to them,
if sip.conf says:
[general]
allow=g729,ulaw
then it always picks ulaw, even though g729 is listed first.
However, if sip.conf says:
[general]
allow=g729
then g729 is chosen.
Dial([EMAIL
John Novack and Time Bandit,
Thank you for your excellent advice and for correcting me on the 12V
power connector issue. I feel confident to move forward on this project
now.
Thanks,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
Jean-Michel Hiver a écrit :
Hi List,
I am looking for a 1 FXO analog termination device, other than the
obvious PC + FXO card, and which can achieve decent call quality. The
SPA-3000 seems an option... have you got any other ideas?
Tiger G104 has PSTN to VoIP and vice versa. Didn't had time
*I suggest the digitnetworks fxo oem it has a good cuality of sound and a
good delay*
2006/11/17, Jean-Michel Hiver [EMAIL PROTECTED]:
Hi List,
I am looking for a 1 FXO analog termination device, other than the
obvious PC + FXO card, and which can achieve decent call quality. The
SPA-3000
Hi
I just press * to retrieve the caller again - Have you tried that?
No, I haven't. Thanks, it's perfect for me.
Conrad
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OK
Thank you very much.
On 11/16/06, Alberto Pastore [EMAIL PROTECTED] wrote:
Antonio Almodóvar ha scritto:
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an
Hi Folks,
Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is
voice quality comparable to standard consumer residential 2.4GHz cordless
phones in the US$30 - $50 price range, or better/worse?
How about handset and speakerphone quality for the main phone?
Seems
Hello,
From some days ago, when i made changes in web interface to asterisk
that comes with trixbox (freepbx), this dont reflect the changes in
asterisk configuration.
I has reviewed the file permissions in /etc/asterisk and all files are
writable to asterisk user.
In freepbx all appears to be
Scott - I find the phone to be a great product, as it stands today with
current firmware. I use it extensively for customer deployments, and
everyone seems really pleased with the features and performance. We
have hundreds of these in the field. I much prefer DECT to WIFI for
client sites where
We have had good results mostly from this unit except for one issue that is
currently being looked into by Aastra. The issue is if a second call comes
in and the cordless answers then puts the call on hold audio drops one way
on the handset. Aastra was able to reproduce this and is working on it.
It is unclear to me if the metermaid patch should be in 1.2.13 or not.
Please advise.
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
TESCO Group Companies
Fax. 248-836-5101
www.TESCOGroup.com
Board member of
www.glimasoutheast.org
Scott,
I've used the phone for 9 months. It's a truly outstanding phone. The cordless
handset sounds great. It is limited to two ongoing calls at one time, but that
has not been an issue for me.
The range on the cordless is comparable to the Panasonic KX-TG4000 KSU that I
used to use and a
With FreePBX you can not modify certain conf files - many are
overwritten at reload
Bart
Pedro Silva wrote:
Hello,
From some days ago, when i made changes in web interface to asterisk
that comes with trixbox (freepbx), this dont reflect the changes in
asterisk configuration.
I has reviewed
Hi Pedro,
Did you press the red bar at the top of the page? Until you do this, the
config files are not written out.
Alex
On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote:
Hello,
From some days ago, when i made changes in web interface to asterisk
that comes with trixbox (freepbx), this
What would be the simplest way to retrieve information form a CNAM
database that provides http based query responses?
Does an application or script already exist that does this?
Basically, I want to do a wget of a URL that contains the callerID
number as a variable, and assign the returned
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 13:32:53 -0700
Subject: [asterisk-users] wget from within asterisk?
What would be the simplest way to retrieve information form
a CNAM database that provides http
hi,
On Fri, 2006-11-17 at 13:32 -0700, Damon Estep wrote:
Basically, I want to do a wget of a URL that contains the callerID
number as a variable, and assign the returned text to another variable
which can be used to set the caller ID name.
agi is your friend.
Matteo
I saw CURL, but it does not register appear in show functions or show
applications, deprecated or add-on?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ
Beaupre
Sent: Friday, November 17, 2006 1:37 PM
To: Asterisk Users Mailing List -
Make sure the curl library/package is installed, then re-compile asterisk.
We're using it on 1.2.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 13:40:40 -0700
Thx!
I saw a note about Curl vs. CURL, is there a difference?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ
Beaupre
Sent: Friday, November 17, 2006 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
They both seem to work, but the Curl spits out warnings about being
deprecated. Ours are all configured using CURL.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov
Do the zaptel drivers need to be told NOT to use the software echo
canceller when using a TDM2400p card with hardware echo canceller, or
does the driver figure this out by itself?
Thanks,
Andrew
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I will be out of the office from November 17, returning November 27.
Mark Stayt
Director of IT
Ocean Optics, Inc
+1-727-733-2447 (Phone)
+1-727-733-3962 (Fax)
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To
Thanks a bunch, this seems to be a simple solution, I just did not have
CURL installed before I built asterisk.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ
Beaupre
Sent: Friday, November 17, 2006 2:05 PM
To: Asterisk Users Mailing List
Options I am aware of for installing curl are yum install in FC4 or
download from curl.haxx.se, neither option distinguishes between curl
and CURL, can someone offer me the slap in the head I need?
Damon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Here is my Extensions.conf file (Default Context). When an
individual calling in dials the extension, the response time seems
very slow. It doesn't immediately go to the next step, but hangs out
for a few seconds (silence)... Suggestions?
Thanks in advance... /pj
[default]
exten =
The Curl/CURL is an asterisk dialplan distinction.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 15:06:49 -0700
Subject: RE: [asterisk-users] wget from within
A quick google search says there isn't anything written yet.
But looking at the database itself, it seems pretty easy to import data into
a sql table or do xml pulls from them directly..
https://www.donotcall.gov/FAQ/FAQBusiness.aspx#download
Webster, Andrew wrote:
Do the zaptel drivers need to be told NOT to use the software echo
canceller when using a TDM2400p card with hardware echo canceller, or
does the driver figure this out by itself?
If I'm recalling correctly, if the drivers see the hardware E.C. it'll
use it
On version 1.2.12.1 running on FC4 with curl.i386 installed the asterisk
CURL function is not registered, perhaps in need something else
(curl-devel.i386 ?)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ
Beaupre
Sent: Friday, November 17,
Phil Jackson wrote:
Here is my Extensions.conf file (Default Context). When an individual
calling in dials the extension, the response time seems very slow. It
doesn't immediately go to the next step, but hangs out for a few
seconds (silence)... Suggestions?
[default]
exten =
It wouldn't be hard to code up at all actually... a little perl magic
and
voila. ;)
Who needs a weekend project?
The Perl magic would be easy. Writing the check to pay for all of that
data is what is so hard...
-MC
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Grettings!
The 0.2.1 version of Destar has been released. Destar is a simple
web-based interface to manage Asterisk. It supports different types of
trunks and phones, many asterisk applications, and Vitual/hosted PBXs.
It can be downloaded from:
http://destar.berlios.de/
Or directly from:
I'm surprised someone doesn't come up with a consortium for all the
asterisk users to poll a central location or does the data come with
restrictions about sharing the data?
Duane from e164.org says he's already built the application you are
looking for to deal with Australian databases if that
Depending on your organization, you're allowed up to 5 area codes for free.
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] ]
Sent: Friday, November 17, 2006 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
I need to ring a group of 8 phones, but not if they are already on
another call. How can I determine which of those 8 phones are busy so I
only ring the others?
--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
+52-55-91169161 ext 2001
Carlos Chavez wrote:
I need to ring a group of 8 phones, but not if they are already on
another call. How can I determine which of those 8 phones are busy so I
only ring the others?
chanIsAvail
Steve
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Alright, I've figured out that by adding a wait to the dial I can get it
to connect to the inter-tel pbx. I still can't get it to either a) pass
the caller id, or b) talk to the correct extension. The inter-tel box
always redirects the call to the operator.
When the call comes in on the T1
Hi,
2006/11/17, Alex Robar [EMAIL PROTECTED]:
Hi Pedro,
Did you press the red bar at the top of the page? Until you do this, the
config files are not written out.
Yes, i press the red bar and freepbx dont return any error.
For example, If i add a new extension, the files
Oddly enough, there's really nothing stopping one from doing so in the
material I just scan through at:
http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm
http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm
In regards to the fee, here is the latest:
The amended rule
On Fri, 2006-11-17 at 16:03 -0800, Steven Ringwald wrote:
Carlos Chavez wrote:
I need to ring a group of 8 phones, but not if they are already on
another call. How can I determine which of those 8 phones are busy so I
only ring the others?
chanIsAvail
The problem with
Hi,
Does anybody know how exactly the jitter buffer in the zap channels
work? Is it adaptive or fixed?
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
jitterbuffers=30
This setting puts 600ms of jitter buffer but the call does not sound as
if it had a .6 second
You can't do any modifications in extensions_additional.conf and
sip_additional.conf. Same is true for extensions.conf and sip.conf, and
other original trixbox files. As soon as you press the red bar, they are
returned to their original state. For modifications, you create your own
files or use
Aastra is a great phone for sound quality and other features. I didn't have
any problems with it and didn't go back to Grandstream once installed
Aastra. My only concern was some problem with its web UI bugs, but that will
be eventually fixed.
___
Its Cisco. Please disable VAD and voice compression in your Cisco equipment.
I had exactly the same problem which haunted me for more than a year and I
tried everything, asked everyone, and no one could solve the problem, until
the service provider told me they had some voice compression feature
You can use SIP_CODEC variable. Read README.variables file.
On 11/18/06, Mark Price [EMAIL PROTECTED] wrote:
Hi,
My pstn provider is currently set up so that when asterisk sends an
outbound SIP call to them,
if sip.conf says:
[general]
allow=g729,ulaw
then it always picks ulaw, even though
I think you guys are all misunderstanding the problem here. Unless I'm
misunderstanding, Pedro's issue is that when he makes changes in FreePBX,
those changes are not written out to the config files.
Alex
On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
You can't do any modifications in
bails wrote:
Ronald Wiplinger wrote:
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There
is a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots of
Is it possible to strip the plus sign from the ${EXTEN} var on an incoming call?
We have our system setup to deal with incoming calls to numbers
without a plus sign, lots of AGIs and databases we don't want to have
to change.
We have seen things like this ${EXTEN:1} which you can use in the
We ran into a Beta version of FreePBX a few weeks ago that was doing
this.. So, if you are running a beta version, upgrade or downgrade and
see if that does the trick.
On 11/17/06, Alex Robar [EMAIL PROTECTED] wrote:
I think you guys are all misunderstanding the problem here. Unless I'm
voiplist wrote:
Is it possible to strip the plus sign from the ${EXTEN} var on an
incoming call?
We have our system setup to deal with incoming calls to numbers
without a plus sign, lots of AGIs and databases we don't want to have
to change.
We have seen things like this ${EXTEN:1} which you
Does anyone have a copy of spc.exe they could send me?
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