Tzafrir Cohen wrote:
On Mon, Oct 10, 2005 at 11:10:59AM -0400, Kanuri, Seshu (Company IT) wrote:
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL.
Hi
Broken into paragraphs, so people can actually read. I will not address
the content, as it has been rehashed ienough here and elsewhere.
Please read, e.g. http://perens.com/Articles/Economic.html before
posting anything in reply to this thread or anything similar.
On Mon, Oct 10, 2005 at
On Mon, Oct 10, 2005 at 11:10:59AM -0400, Kanuri, Seshu (Company IT)
wrote:
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL.
It depends who
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
yes We checked. is not included.
On 10/10/05, El Flynn [EMAIL PROTECTED] wrote:
oner asterisk wrote: Hi all,I would like to add indication tones ,What I did is
enter data to zonedata.c and indications.conf than compile zaptel. and restart asterisk.But it's not working what else I should do
Long list of questions to follow:
Short version:
Does the register line mate to a peer or is a register line totally
unrelated to a peer that is defined?
When using a register line does it have to refer to actual hostnames?
or can you refer to the peer name in the register line instead
of an
At 15:22 10/10/2005, Anders Svensson, wrote:
Hi!
I have problem with my AAH. I have set up a sip channel. It works perfect
both ways with one exception. When someone calls in I only get 1 signal.
The caller have normal ringtone until message is played. Anyone who can help?
Perhaps set
And how exactly is Asterisk relevant to a CMS? could you give a more
specific example?
This is relevant where Administrative users wanted to manage their
Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc
Seshu
NOTICE: If
We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get
over 500 frame errors and over a 500 slip errors per hour. When the errors
reach 1000 per hour the Mitel will take it's T1 card offline. At that point
no calls can be routed from the Asterisk server to the Mitel and the
Folks,
I've been trying to handle the problem where
blocked callerids appear as coming from
asterisk asterisk
on the email notification, and the message
envelope simply doesn't say anything (does not
actually play the vm-unknown message).
So, following the tip provided by several
previous
hello,
I have been using asterisk now for about 2 years now on a RH8.0 it is our
main call gateway.
I have on the box 3 T1 TDM cards connected to 2 Rhino channel
banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA
186s.
It has been working good till today some few hours
what does this mean and how do I fix it?
Channel 'SIP/3044-80e1' sent into invalid extension
'573486' in context 'redial-from-local', but no
invalid handler
Here is the
[redial-from-local]
_91NXXNXX,1,Macro(redial,${EXTEN})
_91NXXNXX,2,Congestion
I use to do the same thing with an SX200 digital, and had the same problem.
I could not get the slips to go away no matter what I tried.
My sx200 would go offline once a day in the middle of the night luckily
once the limit was exceded.
But I did find that down in the t1 parameter settings you
We have 4 employees and were running Cisco 7970
phones. Each phone has a unique SCCP line configured (in the autologin
area of the sccp.conf file) for each employee. We have hints set up in
the extension.conf file like the following:
exten = 101,hint,SCCP/101
exten = 102,hint,SCCP/102
On 10 Oct 2005, at 19:46, lenz wrote:
In data Mon, 10 Oct 2005 14:03:55 +0200, tim panton
[EMAIL PROTECTED] ha scritto:
Yep, I'm working on such a thing.
I have a demo version running at http://www.westhawk.co.uk/
software/faceless/CallUs.html
You don't even need to install it, it runs
hi all expert,
I am testing asterisk like small sip server, i installed asterisk in debian.
It runs very well. I can use softphone to register, but each time i have
modify the sip.conf, i find that it's not good way.
So if i understand, avec asterisk version 1.2 i can use mysql to stock
the
At 16:55 10/10/2005, julien bos, wrote:
hi all expert,
I am testing asterisk like small sip server, i installed asterisk in debian.
It runs very well. I can use softphone to register, but each time i have
modify the sip.conf, i find that it's not good way.
So if i understand, avec asterisk
I forgot to state that this is only for INCOMING
calls. I'm not making outgoing calls, so I really
don't care what the outgoing caller id is.
I'm running Asterisk 1.0.5 stable ... it's a
production environment, and the users are getting
really confused about the caller id strings on their
Geoff Manning wrote:
We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get
over 500 frame errors and over a 500 slip errors per hour. When the errors
reach 1000 per hour the Mitel will take it's T1 card offline. At that point
no calls can be routed from the Asterisk server
Hi Beonice,
Just told i was dealing with the same problem.
[incoming]exten = _!,1,GotoIf($[${CALLERID} = unknown]?2:5)exten = _!,2,Set(CALLERID(name)=Withheld Number)exten = _!,3,Set(CALLERID(number)=00)exten = _!,4,Goto(8)
exten = _!,5,GotoIf($[${CALLERID} = asterisk]?2)exten =
On 10/8/05, Rich Adamson [EMAIL PROTECTED] wrote:
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. show translations verifies that the registration
took place.
When I place a call, having allow=g729 as the only allow option in
iax.conf, I get the
Chris,
Thanks also for your response... That is one of the first things I checked...
I have checked, double checked, triple checked my account with teliax,
and made sure that the g729 box is checked for both sip and iax.
Thanks again,
JR
On 10/8/05, Chris Coulthurst [EMAIL PROTECTED]
Hi Tzafrir! I didn´t know that the last version of saynumber would do
that. I´m having problems with brazilian portuguese that needs an and
sound after de tens and units for numbers higher than 20.
Just like:
twenty and one
forty and two.
Best regards,
Ricardo Poppi.
Geoff Manning wrote:
We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get
over 500 frame errors and over a 500 slip errors per hour. When the
errors
reach 1000 per hour the Mitel will take it's T1 card offline. At that
point
no calls can be routed from the Asterisk
OK, I have just discovered what may be a conceptual flaw in realtime
extensions.
DB schema as follows:
uniqueid, context, exten_id, priority, application, application_data
Primary key is uniqueid with an compound index of context and exten_id
data looks something like this:
x, sip,
--- Dan Journo [EMAIL PROTECTED] wrote:
Hi Beonice,
Just told i was dealing with the same problem.
[incoming]
exten = _!,1,GotoIf($[${CALLERID} =
unknown]?2:5)
exten = _!,2,Set(CALLERID(name)=Withheld Number)
exten = _!,3,Set(CALLERID(number)=00)
exten = _!,4,Goto(8)
exten =
You may have already tried this, but in the past whenever slips come into
the picture on my T1s, crimping a new end for the CAT5 cable seems to help.
We run T1s to a 110 block. Every once in awhile, the 110 needs to be
repunched.
I have found that slips can clear up when we rerun the
I just purchased credits with voipbuster to use it with asterisk. I have not been having any problems, however I'm curious if they have a 1 hour limit on calls?? I have used it to join bridge calls for work, and i got dropped at the 1 hour mark. has anyone else experienced this? its not really
On Mon, 10 Oct 2005 19:27:51 -0500, Justin Richards wrote:
I just purchased credits with voipbuster to use it with asterisk. I have not
been having any problems, however I'm curious if they have a 1 hour limit on
calls?? I have used it to join bridge calls for work, and i got dropped at
the 1
Below is my try of installation to the latest CVS. I have not updated 3
months. What do I miss since last time?
bye
Ronald
/usr/local/src/asterisk # make clean; make update; make install
build_tools/make_version_h include/asterisk/version.h.tmp
if cmp -s include/asterisk/version.h.tmp
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. show translations verifies that the registration
took place.
When I place a call, having allow=g729 as the only allow option in
iax.conf, I get the following error:
WARNING[361]:
Hello all,
yes there is a lot of information about this on the wiki and in past posts on
this list but have not found anything that has solved my problem.
setup is:
phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is
inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only.
I have been battling this problem for 2 months with no resolution as of
yet with TelaSIP. I am told that it is a provider problem(Level
3) because all TelaSIP is doing is passing the call directly to them
once the call comes through.
Anyone else having this issue with TelaSIP or Level3?On
We had this problem a few months ago but they resolved it for us. I
really don't remember more than that.
Darren Wiebe
[EMAIL PROTECTED]
Tom Vile wrote:
I have been battling this problem for 2 months with no resolution as
of yet with TelaSIP. I am told that it is a provider problem(Level
OK, I'm starting to get somwhere with this, I'm at least registering now..
however My inbound calls are still coming into the context defined
in [general] of sip.conf and not into the context I have defined
in my peer and extensions.conf
Here is what I have done:
IN sip.conf:
register =
I'd be curious to hear any Podcasts from the upcoming Astricon
conference. If anyone in attendance/organizing the event is going to
be recording any audio please share. Cheers, HJ
___
--Bandwidth and Colocation sponsored by Easynews.com --
Anyone else having this issue with TelaSIP or Level3?
Yes, to some extend. I have had more luck with incoming calls with IAX
from Telasip, but it's still not 100%. On SIP even two digit
extensions would end up with double digits (12 as 112, etc). I
couldn't find a resolution although Telasip has
modprobe wctdm
FATAL: Error inserting wctdm
(/lib/modules/2.6.8-24.11-default/extra/wctdm.ko): Unknown symbol in
module, or unknown parameter (see dmesg)
FATAL: Error running install command for wctdm
ls -l `locate wctdm.ko`
-rw-r--r-- 1 root root 390308 Oct 11 09:37
OK, I'm starting to get somwhere with this, I'm at least registering now..
however My inbound calls are still coming into the context defined
in [general] of sip.conf and not into the context I have defined
in my peer and extensions.conf
Here is what I have done:
IN sip.conf:
Yep, I'm stunned that as a technical social network we're not leveraging
the technology through webcasts/online presentation, dial in conference
calls for the sessions etc.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry
On Monday 10 October 2005 23:10, Dean Collins wrote:
Yep, I'm stunned that as a technical social network we're not leveraging
the technology through webcasts/online presentation, dial in conference
calls for the sessions etc.
We did have dialin conference calls and even IRC up on the big
On Mon, 2005-10-10 at 23:10 -0400, Dean Collins wrote:
Yep, I'm stunned that as a technical social network we're not
leveraging
the technology through webcasts/online presentation, dial in
conference
calls for the sessions etc.
But they charge admission for astricon, who would pay for the
FYI, Im using g729 with Teliax, and have been for about 1 week with no
problems, good audio quality.
They DO seem to drop registrations unexpectedly at times, but as for codec
usage, so far so good.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Rich Adamson [EMAIL
The question is would people choose not to go if it was necessarily
available as a broadcast.
You're thinking old school.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Monday, 10
On 10/11/05 08:50 Ronald Wiplinger said the following:
/usr/local/src/asterisk # make clean; make update; make install
build_tools/make_version_h include/asterisk/version.h.tmp
if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ;
then echo; else \
mv
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both? I have looked at the records
when the call hangs up.
certain calls, I was thinking of doing something with FastAGI so that
On 10/10/05 22:31 Giovanni Barbis said the following:
-- Executing Dial(SIP/222-23da, Zap/1/34844503450||tTH) in new stack
what does your features.conf say ? you're dialing with options t, T and H.
do read up on what those options do to your call.
--
Regards,
On 10/10/05 19:53 Rich Adamson said the following:
If you don't have any T1/E1 connections to the outside world, then
pick one channel bank and call it your official source of sync, and
would other pbx boxen, like the ericsson md110 serve as good timing sources ?
--
Regards,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of El Flynn
Sent: Monday, October 10, 2005 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Clicks, pops and noise
Rich Adamson wrote:
It is on one
Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet
bridge? For example something like a net4801 with a TE110p in it and then
using TDMoE to get it into a bigger server where the call processing proper
will occur.
Anyone know if it might handle a quadspan card ok? (no
On Tue, 2005-10-11 at 00:05 -0400, Dean Collins wrote:
The question is would people choose not to go if it was necessarily
available as a broadcast.
You're thinking old school.
Dean
I am thinking the convention was set up for money, I cant believe that
the rate generates no profit.
The quadspan card isn't a low profile card is it? I don't think it'll even physically fit in the net4801's footprint.
On 10/11/05, Craig Guy [EMAIL PROTECTED] wrote:
Has anyone on the list used a Soekris engineering PC as a TDM - Ethernetbridge?For example something like a net4801 with a TE110p
On Sun, 9 Oct 2005, C F wrote:
Reading the patents and the comments written here, I couldn't resist
Could you provide the patent numbers?
--
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
- -
- Jason Pyeron
Hi,
* queue application delays about 10 seconds to connect
to the agents.
queue.conf
-
[general]
;monitor-format = gsm
[default]
timeout = 4
; How long do we wait before trying all the members
again?
retry=1
; Maximum number of people
On 10/11/05 12:34 trixter http://www.0xdecafbad.com said the following:
Adding the record functionality and muting participants would also mean
that the hub server would be able to make audio files available after
i'd think that muting would be a prerequisite, even if recording was not
done.
I was wondering if anyone has put together a comprehensive list (that is
reasonably maintained) that lists country codes, landline numbers,
mobile numbers, etc. The particular requirement is for a dialplan to
know what is going to be charged to whom.
For example, mobile and landline rates are
On Tue, 2005-10-11 at 12:57 +0800, Dinesh Nair wrote:
On 10/11/05 12:34 trixter http://www.0xdecafbad.com said the following:
Adding the record functionality and muting participants would also mean
that the hub server would be able to make audio files available after
i'd think that muting
On Mon, Oct 10, 2005 at 07:39:21PM -0300, Ricardo Poppi wrote:
Hi Tzafrir! I didn´t know that the last version of saynumber would do
that. I´m having problems with brazilian portuguese that needs an and
sound after de tens and units for numbers higher than 20.
Just like:
twenty and one
On Tue, Oct 11, 2005 at 08:50:56AM +0800, Ronald Wiplinger wrote:
Below is my try of installation to the latest CVS. I have not updated 3
months. What do I miss since last time?
bye
Ronald
/usr/local/src/asterisk # make clean; make update; make install
build_tools/make_version_h
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