On Tue, 3 May 2005, Andrew Kohlsmith wrote:
> On May 3, 2005 02:22 pm, Ryan Courtnage wrote:
> > From what I've read, glare is common in 2-way loopstart (kewlstart)
> > circuits, and is impossible(?) to eliminate completely. But now I'm
> > wondering what Nortel would tell a customer who experie
Hello,
I have 6 Asterisk switches all running together nicely with DUNDi and
have one minor problem with inter switch CODEC negotiation.
I use G729 (licensed from Digium) on several of the switches.
Inbetween the G729 switches we can make calls no problem.
>From a switch that only does ULAW they
I would use g.729, and if this is an issue, GSM.
Setup trunking between both IAX peers so that you can save a lot of
bandwidth.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Wednesday, 4 May 2005 00:52
>
Hi,
> -Original Message-
> I use MRTG to graph Active/Configured SIP channels and
> Active/Total
> PRI/ZAP channels, but I don't monitor the up/down status. You
> could probably
Any chance you will share the mrtg setup you used for that ? How did you
read out asterisk (via manage
May 3 23:10:52 NOTICE[9159]: chan_sip.c:7938 handle_response: Outbound
Registration: Expiry for sip.broadvoice.com is 20 sec (Scheduling
reregistration in 15999 ms)
I'm behind a linksys befsr41
5060 udp is being forwarded and all of that.
sip show peers
Name/username Host
Hello All !,
Tell at somebody it has turned out to make a transfer of a call between 3 sccp
phones.
|---| |---| |---|
| A | ---> | B |-(#)->| C |
|___| |___| |___|
sccp sccp sccp
And if that has turned out as? If that is possible an example of a
configuration.
-
What's the diffeance???
I just logged im and saw the same screens.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent: Tuesday, May 03, 2005 7:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Nufone
Nufone is now finis
Is anybody using NVBackgroundDetect to detect fax signal on SIP protocol
from ATA?
--
#Joseph
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On Tue, 2005-05-03 at 22:01 -0700, Robert Goodyear wrote:
> On May 3, 2005, at 6:46 PM, Matt Riddell wrote:
> > How many other people are there here that write music? Would there be
> > any interest in creating a pool of music for Asterisk?
> >
> Good idea. Might be an interesting niche to fill
On Artisoft PBX systems I used to use a nifty program call IMS Music on
hold (http://www.nch.com.au/ims/)
It would play loops of music and mix canned scripts for voice overs. IT
would allow you to set music on hold messages by time date and
frequency. It is a windows program but it has a free ver
On May 3, 2005, at 6:46 PM, Matt Riddell wrote:
Chris Mason wrote:
Why not?
Because you have not licensed the file for broadcasting across your
telephone network.
How many other people are there here that write music? Would there be
any interest in creating a pool of music for Asterisk?
Would
On May 3, 2005, at 6:32 PM, snacktime wrote:
On 5/2/05, Robert Goodyear <[EMAIL PROTECTED]> wrote:
On May 1, 2005, at 11:39 AM, Gene Naden wrote:
When we call out from our Asterisk system we consistenly lose the
first
roughly 1500 milliseconds of the audio from the destination. This is
easiest
I'm trying to debug why native bridging isn't working for me when
connecting to various providers via iax. Do they disable it due to
not being able to get an accurate cdr? When my end tries to brdige
the call, iax debug shows me sending a TXREQ, and my provider
returning a TXREJ.
Chris
_
AMP does not automatically add the trunks to FOP. You
can read the AMP docs to see how to do this. You can
put in an enhancement request to the AMP folks if you
want AMP to add trunks to FOP automatically.
--- Mike Price <[EMAIL PROTECTED]> wrote:
> Just to let you know, V 1.0 does fix the ZAP cha
I use MRTG to graph Active/Configured SIP channels and Active/Total
PRI/ZAP channels, but I don't monitor the up/down status. You could probably
write a little perl script to tail the logfile and watch for certain events,
then forward them by mail. Actually, I think I might do that too sinc
Are you using asterisk @ home?
- Original Message -
From: "Patrick M. Gray, Jr." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, May 03, 2005 9:22 PM
Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration
I still can't get the "mul
It has something to do with the AGI script. Scroll down!
- Original Message -
From: "Patrick M. Gray, Jr." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, May 03, 2005 9:22 PM
Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration
Nortel and Toshiba and so on help eliminate this by routing outgoing calls
starting from the highest trunk backwards and incoming calls of course start
from the lowest trunk and work upward.
- Original Message -
From: "Ryan Courtnage" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List
I've read on the wiki how you can SNMP monitor an Asterisk machine and
from what I read, you're pretty much monitoring the availability of
Asterisk.
I'm looking for a way to be able to monitor the availability of
individual T1 circuits of my TE410P card. During the storm season, some
of our T1
On Tue, 2005-05-03 at 19:08 -0500, jltaylor wrote:
> In the U.S., its called:
> Inbound Call Operator Screening (ICOS) automatically screens and blocks
> incoming third-number-billed or collect calls, or both, so that callers
> cannot charge these calls to your line.
> It's a databse thing.
> James
I think the proper solution would be to use the proprietary Skype API
for Linux and
create an asterisk extension for it. There is a $1050 bounty on
voip-info.org[1] but i don't think there are any takers for it yet. :(
Another suggestion was "... to either get a Skype compatible ATA or
FXS/FXO ad
Hi Smadi. I have tested the script in my box and seems to be working
just fine. By the way, in any programming language, the way to get the
asterisk environment vars, and in general all the communication is
through STDIN, STDOUT and STDERR
try to give us more info so we can help you
best regards
I had to manually add the lines to the make file in apps/If you read
the patch file there is only like 4 lines you have to add.
Regards,
Chris
- Original Message -
From: "Sahil Gupta" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, May 03, 2005 9:02 PM
Subject: [Asterisk-Users] SPAND
Hello all,
I'd like to mention that we've put together a simple Java-based
application that provides a somewhat point-and-click interface to create
an Asterisk dialplan. You can get to the dialplanner at
http://www.lanvik-icu.com/asterisk/dialplanner/index.php
You can create contexts and extensio
Hello, I'm experiencing some problems while setting up my asterisk PBX.
What I want to get done is that every incoming call to SRV_A must be
routed to inbound context at SRV_B. That works fine actually, the only
thing is that if the called party stays on the phone and doesn't hang up
after the
Hi,
I'm having troubles getting SPANDSP working with Asterisk (for faxes), on
a search of google.. I came up with a few links but the rxfax and txfax
modules wouldn't patch or compile into asterisk
Any hints?
Regards,
Sahil Gupta
VoiceValley
___
As
Install it using [EMAIL PROTECTED] which is the automated install version of
AMP.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Scott Kamp
> Sent: Monday, May 02, 2005 9:28 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Dis
Yeah, so I’m an idiot…subject
should have been ‘MeetMe’ not MOH.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] MOH Core
uses ulaw
cd /usr/src/zaptel
make config
This will install the init scripts to start zaptel when you boot your computer.
cd/usr/asterisk
make config
does the same thing for asterisk. Use asterisk -r to connect.
Dylan.
On 5/3/05, Ben Johnson <[EMAIL PROTECTED]> wrote:
> When I restart my computer, I need
Here is the situation. I've got * installed. I have Grandstream BT-100 with
latest beta firmware installed and Cisco 7960G.
[3710]
; -> Grandstream
context=default
type=friend
username=3710
user=phone
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
qualify=2000
disallow=all
allow=
Atxfer is only available in HEAD not stable.
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Redstone
Sent: Tuesday, May 03, 2005 12:48 PM
To: Asterisk User
Subject: [Asterisk-Users] IAX2 attended transfer on 1-0-6 Stable
Hi Guys
I'm still wre
I’m trying to get Asterisk setup as a conference
bridge. When I originally tried MeetMe, I was using GSM and as the conference
got longer, the delay got worse and worse. From my research, I assumed that it
was because MeetMe uses ulaw at its core, so everything is getting transcoded
twice
I still can't get the "multi-line magic" to happen. When I get the second call,
this is what appears on the CLI.
Any ideas?
Thanks!
Pat
dialparties.agi: Caller ID is not set
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
I know this is a frequent topic on the list. Sorry if this creates more
bandwidth but I couldn't get my specific answer from neither the wiki
nor searching the list.
I've read that a P4 3GHz+ should be sufficient to handle a 4 T1s on a
single CPU machine. I am setting up a proof of concept mach
I'm trying to configure 4 T1s into this board. The T1s work just fine.
However, I have a question about setting up the clock source properly.
3 T1s are from the same carrier and the remaining T1 is from another. I
have a configuration similar to:
/etc/zaptel.conf
span=1,1,0,esf,b8zs
e&m=1,24
sp
Chris Mason wrote:
Why not?
Because you have not licensed the file for broadcasting across your
telephone network.
How many other people are there here that write music? Would there be
any interest in creating a pool of music for Asterisk?
Would there be any chance of creating a GPL exception
On 5/3/05, David <[EMAIL PROTECTED]> wrote:
> Andrew,
>
> Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
> ulaw. 20 calls at the same time... 512 connection can't support that... What
> do u think?
I would try ulaw just to get a basis to work from. Frankly I would
expe
I'm not sure whether this is an issue with my provider or if it's just
an iax issue. I have a DID coming into my asterisk box via iax. When
I call it from a pots line and hangup, my asterisk box doesn't detect
the hangup until the call reaches voicemail at which time I get the
following message:
Id suggest using AMP or the Asterisk Management Portal
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
McBee
Sent: Tuesday, May 03, 2005 4:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Good web interface for the enduser
On 5/2/05, Robert Goodyear <[EMAIL PROTECTED]> wrote:
>
> On May 1, 2005, at 11:39 AM, Gene Naden wrote:
>
> > When we call out from our Asterisk system we consistenly lose the
> > first
> > roughly 1500 milliseconds of the audio from the destination. This is
> > easiest
> > to demonstrate wi
just pay yer money to ASCAAP or BMI to license it.
-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Well, that can be done...
Really should not do that though...
Cheer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
The question is will the hitachi WIP 5000 work when crossing subnets?
Anybody doing it?
thanks,
jerry
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In the U.S., its called:
Inbound Call Operator Screening (ICOS) automatically screens and blocks
incoming third-number-billed or collect calls, or both, so that callers
cannot charge these calls to your line.
It's a databse thing.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[E
Holy crap! You mean someone actually "read" my email?
Thanks Andrew. Wish more people would read emails.
-Matthew
> From: Andrew Kohlsmith <[EMAIL PROTECTED]>
> Organization: Benshaw Canada
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: Tue, 3 May 2005 16:27:56 -0
El mar, 03-05-2005 a las 03:43, Deborah MALKA escribió:
> Hello,
>
> I wanted to know if there is a way to dissplay infos from Asterisk on a
> SIP phone ? Because I know Asterisk is very powerfull, so I'm nearly
> sure that there is a way to do it.
Using XML on Directory.xml and services.xml with
When using phones that are using G.711 codec and the calls are recorded
with "Monitor", when played back the files sound great.
When we use gsm codec at one or both ends of the call, the recorded
files sound very bad. Much worse than the audio sounds during the call.
With the "Monitor" command we
Why not?
Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
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h
Andrew Kohlsmith wrote:
BTW are you *really* saving any time by bastardizing your email so much (ur,
u, bcz)... jeez.
I think they teach that crap in school these days ... kids and their sms
cell phones..
Jonathan
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Ast
You could run an automated session out your speaker/mic to an incoming
fxs circuit but to answer your question - No.
Never heard it happen before.
Cheers,
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili
>
This is possible . my question is that you dial the extension # to the
PBX , the extension # change.
Wells.
On 3/15/05, Michael Sanders <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Ive been searching the lists and cant find the exact solution I need using
> *. I need to route voice channels between tw
Nufone is now finished with upgrading their system and now are accepting
new customers..
now go sign up and quiturbitchin about broadvoice and their lack of.
everything
http://www.nufone.net
:-)
signature.asc
Description: This is a digitally signed message part
__
You Bring up a great point. I understand these codes and my system
brings them in via ss7 but as youself I don't know how to protect my
network from these charges. I will follow this post to see if anybody
has a fix.
Rodrigo P. Telles wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Folk
No. Sorry. That isn't the answer because Asterisk ignores that particular
startup option.
I always access CLI via:
/usr/sbin/asterisk -Rdgnvv
..always..
And I still get color.
-Matthew
> From: Andres <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - No
Easy..
go to the web interface of the Handset you want to modify...
go to admin login
then select advanced
go to the 'phone' tab, and under suplementary Services there is a whole list
of things that you can enable and disable on the phones..
Including DND
Dave
Does anyone know if there is a way to turn DOWN the verbosity of the
Voicetronix channel driver?
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Running CVS-HEAD-04/12/05-16:39:24 on CentOS 4.0 final installation. I am
hearing a brief echo on our Cisco 7960 phones when a incoming call is
answered. After a few seconds of conversation the echo disappears. There is
no echo on outbound calls or transferred calls. After a search of the
maili
> --
>
> Message: 3
> Date: Tue, 3 May 2005 06:40:37 -0700
> From: "Wiley Siler" <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] Re: LiveVOIP
> To: "Luki" <[EMAIL PROTECTED]>, "Asterisk Users Mailing List -
> Non-Commercial Discussion"
> Message-I
On May 3, 2005 04:14 pm, David wrote:
> Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
> ulaw. 20 calls at the same time... 512 connection can't support that...
> What do u think? The only reason I'm using iLBC is bcz of the Bandwidth.
> What about the packet lost, I see s
Wiley Siler wrote:
I highly recommend this isntall
Asterisk at home
http://asteriskathome.sourceforge.net
You have control of passwords so you can restrict some of the access to
the GUI stuff.
It utilizes...
Asterisk Management Portal (aka AMP)
http://amp.coalescentsystems.ca/
The version 1.0 do
On 16:14, Tue 03 May 05, David wrote:
> Andrew,
>
> Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
> ulaw. 20 calls at the same time... 512 connection can't support that... What
> do u think? The only reason I'm using iLBC is bcz of the Bandwidth. What
> about the packet
This is an ordinary HP/Compaq/IBM server. You can install * on those servers
and install CCM on a ordinary computer with Intel chipset without much
problems.
I.N.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Walid Azab
> Sent: Thursday, Apri
Had a problem today when the CEO of the company tried to go into a
meetme extension. This has worked in the past for him and other
employees but for some reason this time it caused Asterisk to crash.
Asterisk: CVS-HEAD-01/17/05-08:45:53
OS: Slackware
Digium HW: T100P
/var/log/asterisk/messages
If there is another MoH source what is the correct way to use it with
extensions? Let me explain it - asterisk has MoH on extension 555.
Call comes on extension 111, so asterisk should connect incoming call to
extension 555 until someone answers on extension 111.
Second question: if there is a tr
On 16:35, Tue 03 May 05, Christopher McBee wrote:
> We are going to be deploying an asterisk/polycom setup at a customer's site
> and have been trying to find an easy to use web based management system for
> asterisk for modifying common settings. I found the webmin plugin but
> unfortunately,
What do you mean?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki
> Sent: Tuesday, May 03, 2005 3:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Is there any chance to bring Skype
> and Aste
When I restart my computer, I need to run ztcfg before running asterisk.
Can anyone help me with a script that will run ztcfg before starting
asterisk ??
Thanks
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Hi,
thanks a lot. After adding username, password and changing
extensions.conf my very basic * setup works excellent.
So, next milestone is establishing connection from my office to my home
server where my * resides...
Thanks and Regards,
Primoz
-Original Message-
From: Bellows, Jar
This was on a DELL 750 P4, 3.2Ghz, with Digi 4 port FXO
voip-gw1:/usr/src/zaptel# ./zttest-mod.o -v
Objective: to read 8192 bytes from TDM card in 1.00 seconds.
Opened pseudo zap interface, measuring accuracy...
8192 bytes in 1.023973 seconds
8192 bytes in 1.023972 seconds
8192 bytes in 1.02397
Chris Mason (Lists) wrote:
> What's the easiest way to handle directory entries on these phones? I
> am using a XML editor and Samba to allow access to the files, but
> it's a bit of a PITA, what is everyone else doing?
I'm building mine out of the company directory, and pre-seeding the
phones.
Only signaling or with media stream also?
You need commercial hardware platform. Those cost ~$20-100K. Probably you
can rent those boxes. I do know, that Spirent Communications has boxes for
SIP/H323/Skinny.
I.N.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED
I highly recommend this isntall
Asterisk at home
http://asteriskathome.sourceforge.net
You have control of passwords so you can restrict some of the access to
the GUI stuff.
It utilizes...
Asterisk Management Portal (aka AMP)
http://amp.coalescentsystems.ca/
W
-Original Message-
F
Hi,
Does anyone know how to tell a cisco 7970 to blank it's screen after
$TIMEOUT?
I can't find an option for the xml config file on google, all i can find
is how to turn it in via cisco call manager, wich I obviously don't
have ;)
Regards,
Joris
___
A
Well, that can be done...
Really should not do that though...
Cheer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Tuesday, May 03, 2005 12:22 PM
To: 'Matt'; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
Just the other day I noticed that my sound on the phone was
turning rather bad, so I had a dig around and when I type tops in linux (RH9) I
noticed that it was running at 99% any ideas?
Regards
Paul Dracevich
Wireless Technology Consultant
Wayby Group
___
Does anyone know of a good way to replace the functionality in ast_readstring
in res_perl? It doesn't seem to be available in res_perl and I'm not sure of a
clean way to get things like passwords, etc.
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Hi. I currently have asterisk set up to log cdr info to my postgres database
and everything is working fine. I was just looking on the wiki and it looks
like there was an project, app_dbodbc, that would allow CID and blacklist
lookups from an ODBC database but the project doesn't appear to be c
I get this problem when I dial out over my voice T1 but not when I dial out
over a POTS line. So it looks like in my case it is the voice T1 provider.
- Original Message -
From: "Robert Goodyear" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Mond
I wrote:
> In article <[EMAIL PROTECTED]>,
> Rich Adamson <[EMAIL PROTECTED]> wrote:
> >
> > It would be very interesting to see everyone's results in running
> > this, and even more interesting to report the results with the OS
> > distro in use, mobo in use (if known), etc. If anyone actually
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Folks,
Does someone knows how to identify and block collect calls on Asterisk using PRI
channels?
I googled it and found this:
http://lists.digium.com/pipermail/asterisk-dev/2004-November/007500.html
I don't know what does it mean!!!
Can someone help
Ryan Courtnage wrote:
> On 3-May-05, at 10:34 AM, Eric Wieling aka ManxPower wrote:
>> It's called "glare".
>
> Thank you, I'm now walking down the right path.
>
> From what I've read, glare is common in 2-way loopstart (kewlstart)
> circuits, and is impossible(?) to eliminate completely. But n
Does anyone have a comprehensive list of all the ways someone can be a
member of a queue(Static, dynamic, agent, non-agent etc) and the
advantages and disadvantages? Seems there are some bugs that are
considered features when using agents(like no SIP transfer) and other
odd problems.
Someone
pinchien wrote:
> What is Asterisk GUI architecture acturally? I could not get it...
>
hmm?
check [EMAIL PROTECTED] - it contains AMP - http://asteriskathome.sf.net
Tomek
--
Startuj z INTERIA.PL! >>> http://link.interia.pl/f186
On May 3, 2005 02:22 pm, Ryan Courtnage wrote:
> From what I've read, glare is common in 2-way loopstart (kewlstart)
> circuits, and is impossible(?) to eliminate completely. But now I'm
> wondering what Nortel would tell a customer who experiences glare on
> their new Meridian system... they mus
Andy,
Thanks for ur reply... Yes.. ServerA has a T400 card (wct4xxp) installed but
serverB got nothing.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton
Sent: Tuesday, May 03, 2005 1:34 PM
To: Asterisk Users Mailing List - Non-Com
Ryan Courtnage wrote:
On 3-May-05, at 10:34 AM, Eric Wieling aka ManxPower wrote:
Ryan Courtnage wrote:
Hello all,
Everyone has probably experienced this at some point in the past:
You pick up your analog phone. Rather than hearing dialtone, you
are connected with someone who has just called yo
> Really no one is to "blame"
> This is known as Glare, or a head on ( collision )
> Take a basic Telephony course before attempting to become a telecom
> engineer.
>
> Back in the "good old days" a PBX would have analog trunks that were
> ground start, and tip was open when idle. The PBX would
The accoustic guitar collection here is pretty nice...
http://www.freeplaymusic.com/search/category_search.php?t=v&i=41
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, May 03, 2005 10:59 AM
To: Asterisk Users Mailing List - Non-Comm
Andrew,
Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
ulaw. 20 calls at the same time... 512 connection can't support that... What
do u think? The only reason I'm using iLBC is bcz of the Bandwidth. What
about the packet lost, I see some packet lost... What is the best
If you're going through a CLEC for your lines, they can probably set the
Glare Preference to be You or the Telco. I'm not sure if the Baby Bells
would add that preference option for you.
-m
On Tue, 3 May 2005, Eric Wieling aka ManxPower wrote:
Ryan Courtnage wrote:
Hello all,
Everyone has probab
On May 3, 2005 02:58 pm, Andres wrote:
> I think this answers your question:
>-n Disable console colorization <=== use
> this option
I don't think that answers his question at all. I think he's asking why
asterisk is explicitly setting the background colour.
-A.
__
Hi
Luca,
I
am trying to implement your solution for outbound calls through Manager API and
I get OutgoingSpoolFailed. Did it work for you? Can you give me more details
about your example?
Thanks
a lot,
Lior
___
Asterisk-Users mai
On 5/3/05, Michael Bielicki <[EMAIL PROTECTED]> wrote:
> chek your pridialplan setting a well as the prilocaldialplan setting.
Also if you set it as dynamic be aware that it actually strips off the
international and national prefixes, and only tag numbers as
international or national !
___
>From : http://www.techfest.com/networking/wan/t1.htm
T1 has a number of other defined alarm and control signals. The alarm
signals have different color designations and are used to indicate
serious problems on the link. These alarm signals are defined as:
Red Alarm
This is a local equipment al
What is Asterisk GUI architecture acturally? I could not get
it...
___
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Command line as headless Linux.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
pinchienSent: Tuesday, May 03, 2005 12:25 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk
GUI
What is Asterisk GUI architecture acturally? I could not get
it...
__
Hi team,
Basic question I know, but I can't seem to find any obvious information
about this:
Does anyone know if * natively supports call forwarding from a given
extension (i.e. call forwarding without having to write a macro)?
My user wants to be able to dial a code plus a phone number to star
On Tue, 2005-05-03 at 14:15, Christopher Jacob wrote:
> Josiah Bryan,
> Any useful results from your "number of installed systems survey"? If so,
> could you email them to me off-list?
actually, could you e-mail them on-list (might be more appropriate for
Asterisk-Biz though).
__
We are going to be deploying an asterisk/polycom setup at a customer's site and
have been trying to find an easy to use web based management system for
asterisk for modifying common settings. I found the webmin plugin but
unfortunately, that doesn't seem to work.
___
Remco Barende wrote:
Then when I try to start asterisk I get this error:
chan_zap.so: load_module failed, returning -1
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
May 2 20:41:58 WARNING[8663]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
[EMAIL PROTEC
> Can you point me to some basic doc on how to do that? I'm rather
> familiar with linux (about ten years worth), but have never tried to
> flop kernels like that.
First time I compiled my own kernel I followed the instructions on
this page : http://www.voip-info.org/wiki-Asterisk+Zaptel+Installati
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