On 2006-09-23 12:43:32 -0700, Kevin P. Fleming [EMAIL PROTECTED] said:
- Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Also, are you referring to newer ones than the 1.4 downloads that
were
available a couple of days ago or do you mean people running the 1.2
versions?
The versions that were
On Tue, Sep 26, 2006 at 08:43:11PM -0700, Nick Ellson wrote:
The link is not working at OpenVox.
There's a download link in the bottom of the page, that leads to:
http://www.openvox.com.cn/members_downloads.php .
That page has the A1200P device driver as a download item (not just
for
Hi,
I'm struggling with this kind of problem:
my hardware sip phone is registering to Asterisk 1.2.10 successfully,
but when I send INVITE to server - it receives the packet but (in sip
debug mode) I see: 'Ignoring this INVITE request'.
While searching in 'chan_sip.c' I've found that this
Hi. I'm having a bit of trouble with outgoing calls on zap channels.
When i try to make an outgoing call asterisk doesn't detect if the other
party answers. When i run 'show channels verbose' in CLI asterisk tells
me that the respective channles are in ringing state like this:
Channel Context
Hi List!
Is there any way to set outgoing CID number when making VoIP calls using VoIP
Buster? I have search on their forum and I couldn't find anything useful. There
is no support mail on their web pages :((
P.S.
I use them because they are cheep and sound quality is satisfying
--
Tomislav
Hi, I installed this beta and I'm trying to use the jingle integration, following the steps in this wiki
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk,
but I'm having some problem. I registered even a SIP than a IAX user;
when I try to call the jingle user connected via
Hi everyone, I need to use Asterisk 1.4-beta2
due to its jingle compatibility, but I've read that there are some
modules issues upgrading from a previous version. How can I remove a
previous version to have a clean install?
___
--Bandwidth and Colocation
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have one extension that rings in many places. It has just come to my
attention that I can only monitor 4 devices within a hint.
Ex:
exten = 132,hint,SIP/DEVASIP/DEVBSIP/DEVCSIP/DEVD
if I add SIP/DEVF, DEVF is not monitored.
I'm
- Original Message -
From: Tomislav Parčina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 27, 2006 8:44 AM
Subject: [asterisk-users] Voip Buster - CID
Hi List!
Is there any way to set
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
No. I once tried to create a channel variable during hangup. Then, in
the hangup extension this variable was added to the user defined CDR
field. This generally works, but only if the call leg hangs up, on which
the AOC is received. In
I'm interested, why do you monitor multiple devices within a hint? If one device is in use (and three are free), how does it show - in use or as free?
I'm glad you asked :-) If we had Shared Line Appearances, I would not have to do this. However, I could be at any of about 6 different phones,
Hi,First, Thanks a lot for your help.My responses are in your mail:2006/9/27, Frederico Madeira [EMAIL PROTECTED]:
Nicolas,
We use a TE110P from digium. We make the same procedures oriented in that website. the only change was in signaling as i've said previously.We try pri_net and
Ok, I just setup a test setup that allowed for five devices (actually in this case five lines) to be monitored.
Next question, does anyone know if there is a limit to the number of characters allowed for the hint? That may be what's causing the issue. I just switched to using the MAC addresses
Greetings all,
I have a problem with a PBX that I manage. The system has 2 AVM Fritz boards
connected to two BRI ISDN services using chan_capi in addition to several SIP
trunks going out to Internet based providers for call termination via the
Internet.
They experience problems when the
On 07:10, Wed 27 Sep 06, Ronald Wiplinger wrote:
I want to make the context [default] as an alarm, for not having
set-up correct.
I am looking for a way to get incoming calls via ENUM or via names (e.g.
sip:[EMAIL PROTECTED]) into a defined context. How can I do that?
If you find out
Melcon Moraes wrote:
What a confused message, isn't it?
As far as I could understand, if you're getting a RJ45 for conection,
you won't need any kind of adaptor. For coaxial cable, you'll need a
balun. That's all layer 1 talk - physic layer
Yes, you need to know a lot more about your pbx
Hi,
At our other site in the UK we currently have a rather old Nortel BCM (4000 I
think), with an ISDN30 feed and 15-16 or so digital extensions (Meridian of
some description).
The ISDN comes in as HSDSL over a twisted copper pair to a small BT box, then
ethernet to the BCM.
We'd like to do,
You can interface between the digital phones and an Asterisk machine
using a Citel SIP Handset Gateway from www.citel.com. The sales
department on +44 (0)115 940 5444 will be able to give you some pricing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On 27 Sep 2006, at 10:56, Mike Williams wrote:
Hi,
At our other site in the UK we currently have a rather old Nortel
BCM (4000 I
think), with an ISDN30 feed and 15-16 or so digital extensions
(Meridian of
some description).
The ISDN comes in as HSDSL over a twisted copper pair to a small
It depends on the actual given environment, but you could also think
about using Linksys' PAP2 adapter!
mike wrote:
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
I would think channel banks - T1s - TDM card in asterisk server would
work better than a bazillion ata adapaters
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Artner
Sent: Wednesday, September 27, 2006 8:53 AM
To: Asterisk Users Mailing
I am setting up an asterisk box , my first with PRI T1
interface to a Nortel 61C. We have quite a bit of experience with the 61C
and do most of the programming including maintaining several other PRI
interfaces in this switch. The problem we are having is as soon as we
turn up the PRI, on
I am in the process of learning my A1200P, and i would like an elegant way
to prevent it from answering the phone, but still make outbound calls. I
tried zap destroy channel 1 (which worked, but pissed off Asterisk ;)
Is there a more elegant way to tell it to answer/not answer on command?
Hi everybody!
I have some Linux experience but I'm completely new to asterisk.
I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk
(1.2.12) preinstalled and some basic configuration (Wiht a few
extensions). Now I want to implement something more, fox example
voicemail (storing
Norbert Zawodsky wrote:
Hi everybody!
I have some Linux experience but I'm completely new to asterisk.
I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk
(1.2.12) preinstalled and some basic configuration (Wiht a few
extensions). Now I want to implement something more,
Ronnie Jones wrote:
I am setting up an asterisk box , my first with PRI T1 interface to a
Nortel 61C. We have quite a bit of experience with the 61C and do most
of the programming including maintaining several other PRI interfaces in
this switch. The problem we are having is as soon as we
On Tue, Sep 26, 2006 at 08:11:09PM -0400, Kristian Kielhofner wrote:
But gratuituously making easy something that very few people have a
legitimate need to do, which undermines something that -- even if you
do only make the resaonable assumption that you know which phone, and
not which person,
On Tue, Sep 26, 2006 at 09:30:04PM -0400, Kristian Kielhofner wrote:
Steve Totaro wrote:
I set caller ID to a unique identifier before sending to a transfer
partner or overflow call center. This makes it much easier to match
CDRs and get stats on the outcome of calls once they leave our
On Wed, Sep 27, 2006 at 10:21:31AM +0530, Benjamin Jacob wrote:
Jay R. Ashworth wrote:
On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote:
hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan,
/
hello
On Wed, Sep 27, 2006 at 12:20:02AM -0500, Lacy Moore - Aspendora wrote:
I didn't see it as making fun of anyone. I, for one, was curious about it.
I suspected it was some type of translation issue, whether it was a word in
another language that doesn't translate or what. I know there
Is there a more elegant way to tell it to answer/not answer on command?
Put your Zap line in a context that do just this :
s,1,Hangup()
hth
___
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asterisk-users mailing list
To UNSUBSCRIBE or
Nick Ellson wrote:
I am in the process of learning my A1200P, and i would like an elegant
way to prevent it from answering the phone, but still make outbound
calls. I tried zap destroy channel 1 (which worked, but pissed off
Asterisk ;)
Is there a more elegant way to tell it to answer/not
Hi All
Would someone be kind enough to provide/point me to a resource when I
can see an example dialplan for making outgoing calls.
All our calls with go out via an ISDN30 gateway so ideally the diaplan
needs to be able to deal with the following errors:
no free channels
user busy
user
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: 27 September 2006 15:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] I doubt it...
The issue is idiomatic usage. I've always assumed
Just wondering if there are any IAX phones worthy of the name phone out
there -- looking for hard phones, but I suppose a Linux-based softphone
wouldn't, you know, hurt. ;-)
Thanks!
-Ken
___
--Bandwidth and Colocation provided by Easynews.com --
Hi,
I have got a Digium TDM04B card (4 FXO modules installed) and i'm having
problems getting it working.
ztcfg reports the following:
asterisk:~# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart
What is wrong with using the WaitForRing app?
Rich Adamson wrote:
Nick Ellson wrote:
I am in the process of learning my A1200P, and i would like an elegant
way to prevent it from answering the phone, but still make outbound
calls. I tried zap destroy channel 1 (which worked, but pissed off
Jay R. Ashworth wrote:
On Tue, Sep 26, 2006 at 09:30:04PM -0400, Kristian Kielhofner wrote:
Steve Totaro wrote:
I set caller ID to a unique identifier before sending to a transfer
partner or overflow call center. This makes it much easier to match
CDRs and get stats on the outcome of calls
Ken - the IAX compatible phones I have seen, for the most part, are OEM
looking, and overall pretty cheaply made.
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax -
Zap channels consider the call answered when dialing is complete, at
least with the analog interface. There is no answer supervision provided
to the PSTN with a POTS line
Don't know if this extends to a PRI or not.
John Novack
Alexandru Voinescu wrote:
Hi. I'm having a bit of trouble with
On Wed, Sep 27, 2006 at 09:12:31AM -0400, Bill Gibbs wrote:
I would think channel banks - T1s - TDM card in asterisk server would
work better than a bazillion ata adapaters
Assuming that you don't need to have a T-1 card in their for your
*trunks*. Since I'm told that you can only have, say,
On Wed, Sep 27, 2006 at 07:45:07AM -0700, Steve Langstaff wrote:
The issue is idiomatic usage. I've always assumed they did it in a
table driven fashion, but I never delved into it.
I have seen quite a few speakers of other languages use doubt in the
meaning of question, inquiry though,
Hi,
Did anyone actually manage setting up a single SER with multiple Asterisk boxes?
I particulary have a problem of keeping the session alive and by that I mean directing
all the following sip messages to the same asterisk box the first signal was sent (randomally).
Please don't direct me to
Just wondering if there are any IAX phones worthy of the name phone out
there -- looking for hard phones, but I suppose a Linux-based softphone
wouldn't, you know, hurt. ;-)
Idefisk looks pretty nice and there is a Linux version :
http://www.asteriskguru.com/idefisk/
There is also iaxcomm :
Jay R. Ashworth wrote:
Assuming that you don't need to have a T-1 card in their for your
*trunks*. Since I'm told that you can only have, say, one Digium card
per chassis, this can be an issue.
You were told wrong. I have had up to FOUR Digium cards in a chassis.
3xTDM400P and 1xTE110P. I
Has anyone actually gotten ASTTAPI to work? I
can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio)
working fine. I have noticed that SNAP and Xtelsio act differently.
Etelescript is the application that will be calling TAPI.
Mike HammettIntelligent Computing
Hello, I have a problem with my X100P card I have connected it to my
asterisk and it works..
but I hear an echo.. I´ve tried echocancelation... echotraining.. and
nothing happens...
I´ve changed the values from the rx and txgain.. from -40 to 10 and it
doesn´t changes anything..
Don´t know
Since I'm told that you can only have, say, one Digium card
per chassis, this can be an issue.
???
lspci | grep Jens
01:01.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537
01:04.0 Network controller: Individual Computers - Jens Schoenfeld Intel 537
asterisk -rx zap show
Using queues here (1 of them), and would like to know
if anyone's written anything like a script that might
tell someone by festival or the like of the status of
a queue, like # of calls waiting, and hold times...
Any other way of finding that out without spending a
ton of money on third party
We got a few 16 ports Media gateway for quite a reasonable price.
Email me for more info.
4 of them and it will end up cost you less than getting channel banks and t1
card.
Sam
-Original Message-
From: Jay R. Ashworth [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 27, 2006 11:55
On Wed, Sep 27, 2006 at 11:15:48AM -0500, Eric ManxPower Wieling wrote:
Jay R. Ashworth wrote:
Assuming that you don't need to have a T-1 card in their for your
*trunks*. Since I'm told that you can only have, say, one Digium card
per chassis, this can be an issue.
You were told wrong. I
Well why pay more when you can get it at much cheaper price.
A single port gsm gateway is around £69 GBP and if you want to know more
info please email me .
Sam
-Original Message-
From: Andrea Spadaccini [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 27, 2006 12:33 AM
To:
Adi,
It is possible to do what you are looking for. It is actually easy.
There is a problem that I have found with ser/openser.. Documentation is
difficult to read and some things are just not there, so you get people
that spend many hours trying to get these functions to work. In these
days
Hi,
I have a client operating a call center in Jordan, he
has a new 5 years
project to make and receive VoIP calls to/from the US.
The project requires a T1 US termination (24 lines)
with at least 99.9%
uptime and perfect voice quality and multiple area
codes.
Can anyone suggest a VoIP
Well, that just makes too much sense.. starting to feel a tad embarrased
here ;) Ok, I will simply remove the Dial(IAX2/4005) and have it not do
anything, that will error on the console, but that's ok and let the
parallel land line have the call (AKA: The wife)
Nick
--
Nick Ellson
CCDA,
Erm.. nothing that I know of, other than I do not yet know what that
means? :)
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Wed, 27 Sep 2006, Eric ManxPower Wieling wrote:
What is wrong with using the WaitForRing app?
Were those people -- who, unlike me, had done it and had problems -- wrong?
There are more variables than the Digium card itself. Things like bus
design, chipset etc all come into play. I've noticed that there is a
concerted effort with Asterisk implmentors to often roll out Asterisk in a
white
Did you do a make make install for add-ons BEFORE doing
so for asterisk?
If so try asterisk first and when all is installed install
add-ons.
--
I tried a make clean make make install for asterisk and then for
asterisk-addons but am still getting the segmentation fault on asterisk
2006/9/26, Steve Underwood [EMAIL PROTECTED]:snipT.38 termination is now fairly solid. T.38 gateway is
also basically working, snipHi,For may understanding, what is the difference between T.38 termination and T.38 gateway ?Regards
___
--Bandwidth and
Nick Ellson wrote:
Erm.. nothing that I know of, other than I do not yet know what that
means? :)
pbx-1*CLI show application waitforring
pbx-1*CLI
-= Info about application 'WaitForRing' =-
[Synopsis]
Wait for Ring Application
[Description]
WaitForRing(timeout)
Returns 0 after waiting at
Hi All,
I have a Linksys SPA-3000 [Hardware version 3.0.0(1178), Software
version 3.1.10(GWd)], with both the FXO and FXS interfaces
registering with asterisk via SIP seperatley. I also have a
Cisco 7940 and 7960 using the sccp2 (chan_sccp) driver, and a
couple of IAX softphones
Both inbound
I am playing with txfax. I have gotten a fax to send which is great.
However now I am creating a multipage fax, I can view all the pages with
viewfax (mgetty-viewfax package)
but when I fax it with txfax I only get 1 page
Any ideas there?
Jerry
I basically do:
gs -q -sDEVICE=tiffg3
I had a problem with the voicemail system hanging after certain users
would enter their password.
I found that lock files get left behind. In order to fix this, in my
startup script I put this line:
rm -f /var/spool/asterisk/voicemail/*/*/*/.lock*
Works nicely. Hope it helps someone
I'm setting up a * system for a friend. Instead of dial 9, he wants
his internal extentions to be prefaced with #.
We have it working on his kid's mac with softphone, his desk with a
gxp2000, but he wants to replace his house phones with two ata-186's .
We have a problem though. The ATA's
I have no experience
on the Nortel side, but will comment on the timing
thingie.
The asterisk T1 card
(port going to the Nortel) will always generate T1
timing on the
transmit side of the T1. There is no way to turn it off
(by T1 Spec's). So,
letting the Nortel use CLOK = EXT is
Hi Zac,
Thank you so much for your sincere answer. What you brought up is exactly
what I encountered when I tried to find a solution for this, the documentation
is inconsistent and ambiguous, and everywhere I look I end up with outdated
examples that make little or no sense in the good case, or
hi thereAdvanced Science and Technology Institute uses Asterisk. On 9/21/06, tubongpeyups [EMAIL PROTECTED]
wrote:hi all,my apologies for posting it here in a technical mailing list. i need some info on companies that support asterisk deployment in the Philippines. Please send me a note
On 9/22/06, Rich Adamson [EMAIL PROTECTED] wrote:
So, it seems there is some type of weird interaction between my system
and the ShoreTel system if I use the SPA941 IP phone.
Does anyone have suggestions as to how I can start debugging this?
Check the RTP Packet Size (under the Sip tab).
Has anyone successfully gotten rpid working between two
phones through asterisk?
Aaron
Daniel
Computer
Systems Technician
Sam
Houston State University
[EMAIL PROTECTED]
(936)
294-4198
___
--Bandwidth and Colocation provided by
We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually).
I'm suspecting their may be some
How do you plan on choosing which Asterisk server to send the SIP requests? Truly random? Based on some sort of LCR methodology?
Have you tried using the LCR module for SER to send the requests to asterisk?
Not sure it would work, but it might be worth looking at.
N.
On Wed, 27 Sep
Hi, for call centers with voip phones and calls coming in via SIP and
Zap, what app_ are people using to do:
-conference
-listening to conversation of agents
Is app_meetme or app_conference?
Does app_meetme still suffers from the need to transcode to slin?
--
Can anyone direct me to a colo provider in the UKwhere I can park an asterisk server and buy UK toll free inbound services over SIP?
Thanks
Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection.
DANIEL, AARON MATTHEW wrote:
Has anyone successfully gotten rpid working between two phones through
asterisk?
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
Aaron,
RPID is supported in Asterisk but many phones do not support
It
won't work, unless you make sure that transfers go through the same asterisk
server as the orignal call went through. Using the SER dispatcher won't fix
that.
-Original Message-From: sip
[mailto:[EMAIL PROTECTED]Sent: Wednesday, September 27, 2006 2:25
PMTo: Asterisk
Adi Simon wrote:
Hi,
Did anyone actually manage setting up a single SER with multiple
Asterisk boxes?
I particulary have a problem of keeping the session alive and by that I
mean directing
all the following sip messages to the same asterisk box the first signal
was sent (randomally).
With a TDM 400 card in the system, WHY do you even need ztdummy??
I thought that was a substitute when there was no other timing source
The only time I have had to compile ztdummy is when there was NO card
present.
Of course, I could be wrong. Please enlighten
John Novack
Eddie Johnson Jr
Ronnie I have 4 non-PRIs connected
to a Nortel 11C and I had played with PRI connections before and got them
working. If you want to call me we can go over your set up and compare with
mine.
Kevin Savoy
701-774-4023
Novo1
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On 15:27, Wed 27 Sep 06, Erick Perez wrote:
Hi, for call centers with voip phones and calls coming in via SIP and
Zap, what app_ are people using to do:
We use SIP and IAX2 and SCCP (chan_sccp). Zap is not
possible for us because we want to run it on OpenBSD and the
zaptel is not ported to it
On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Can anyone direct me to a colo provider in the UK where I can park an
asterisk server and buy UK toll free inbound services over SIP?
Thanks
Probably more relevant on the asterisk-biz list. However I'd be
interested to know what replies
On 27 Sep 2006, at 20:16, Ronnie Jones wrote:
Also, have you tried any of the pri show ... commands in
asterisk, or
any of the pri debug items?
Yes. When the circuit is up I can pri show span 1 and it show
partitioned up and active.
One thing to note - changes to the timing
Yeah... I wasn't really sure. I'm trying to think of a way and nothing comes to mind. The problem is that SER is sort of part stateful and part not, and isn't as concerned with a constant dialog as simply passing the SIP packets effectively. You might be able to couch some logic somehow that
I'm still getting these errors if anyone has any ideas I'd be truly
appreciative.
On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote:
Could the problem is this: Content-Type: unknown?
Reliably Transmitting (NAT) to 192.168.1.228:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP
Hello allI have an asterisk box running Asterisk 1.2.8 and I installed a digium TDM2400 with 8 FXO ports. When I amke a call to the PSTN, the zap channel answers, and teh call goes through if a PSTN is connected to the answered port. However, if there is no dial tone in the answered channel, or if
Be careful when using heavily ChanSpy. We
did couple of weeks ago and the result was having Asterisk crashing
almost once every day. How heavy? around 4 people using it 8 hours a
day, each one using ChanSpy every 3-5 mins.
we were not able to find the exact reason, so just stop using
Hi all-
In 2004, I set up a sms texting process for a UK customer, using the
asterisk SMS command and BT's BT Text SMS facility. This has been
running fine up until recently. A couple of weeks ago, I upgraded them
from their old asterisk version (CVS 2004.07.25) to 1.2.9.1, and have
been
Anyone else having trouble with MWI on 1.4 Beta? The messages are
getting stored and I'm getting the emails but no stutter tone or MWI as
far as I can tell.
MARK.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
I'm experiencing the same problems, but
unfortunatelly haven't been able to associate them with any number
since they appear to be random. But maybe we can do a little research
about it, and hopefully find teh solution for both:
are your PSTN lines POTS or E1/T1? can you make a couple of
Barry D. Hassler wrote:
We seem to be getting unexpected hangups on our * system, very
consistent when calling particular numbers that we can associate with a
clients phone system. These hangups generally occur when our call is
transferred within their system (to voicemail usually).
I'm
Naija Man wrote:
Hello all
I have an asterisk box running Asterisk 1.2.8 and I installed a digium
TDM2400 with 8 FXO ports. When I amke a call to the PSTN, the zap channel
answers, and teh call goes through if a PSTN is connected to the answered
port. However, if there is no dial tone in the
Likewise, Ronnie, we have 2 PRIs going to an 11C - let me know if I can
help.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Sep 27, 2006, at 2:42 PM, Savoy, Kevin - Williston, ND wrote:
Ronnie
I've managed to get asterisk going. For the moment, I simply wish to get a
couple of SIP phones functional.
One is a x-lite softphone, the other a generic hard (sip) phone. Each connects
to asterisk and will give me a dial tone, and accept key input. But neither
can speak to the other, call
I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
depending the busycount param), and call progress will in fact try not
to cut the call due to false hangups.Alyed
Return-Path: [EMAIL PROTECTED] Wed Sep 27 16:12:13
Both can cause random hangups. This is a well known issue. It even
says in the sample configs that these features are prone to false positives.
Alyed Tzompa wrote:
I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
Well, never mind. I seem to have found some docs that may assist.
joe
joe, at j4computers[EMAIL PROTECTED] Wrote on: 9/27/2006 7:22 PM:
I've managed to get asterisk going. For the moment, I simply wish to
get a couple of SIP phones functional.
One is a x-lite softphone, the other a
Hello Folks,
First post. I am using a Trixbox 1.1.1 version and have been working
with it for a few weeks, experimenting and trying to learn. I have
decided to set-up the box as a phone system for a community
organization/club in our area. I have tried to use FreePBX to make all
the
We are looking at putting an asterisks box in place and I was curious to
know who you guys recommend for termination DID's?
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On Wed, Sep 27, 2006 at 12:02:48PM -0600, Colin Anderson wrote:
Were those people -- who, unlike me, had done it and had problems -- wrong?
There are more variables than the Digium card itself. Things like bus
design, chipset etc all come into play. I've noticed that there is a
concerted
which is hard to come by in the closed, secretive telephony world. Tip o' the hat to SHSU. I wouldn't touch *that* install with a space tether.
Has anyone *interviewed* those implementors?Should I go do it?
All you gotta do is say Hey Aaron, how'd you do such and such and I'm sure he'd be more
Verizon.
On 9/27/06, Duracom Lists [EMAIL PROTECTED] wrote:
We are looking at putting an asterisks box in place and I was curious to
know who you guys recommend for termination DID's?
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