On 14:42, Fri 02 Jun 06, Douglas Garstang wrote:
Has anyone got any neat solutions for Asterisk .conf file revision control?
We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_
common set of conf files on. They aren't all the same though. There's subtle
hi
i am experiencing some problems with the configuration of an BN8S0 Beronet card.
I've downloaded last CVS of mISDN ans mISDNuser, i patch the kernel
2.6.16.18 and the enabled the following:
* ISDN support
x x Old ISDN4Linux ---
x x --- CAPI subsystem
x x M CAPI2.0 support
x x
Hello Josué,
we're running Asterisk in combination of the T-Com Octopus E800 with
QSig Protocoll. The protocoll itself is supported but some features are
missing, or i didn't found out yet
how to use them. I'm also interested in how to use qsig for
determinating if other phones are available for
I have no problem with paying Digium the $10 for G729 licenses, everyone
has to make money. It's the administration of the licenses that sucks. I
experiment with different hardware a lot, and make up demo machines to
install for customers with available hardware. I have to put G729
licenses on
Talk to digium about this on [EMAIL PROTECTED], they might be able to
help you out there.
Zoa
Chris Mason (Lists) wrote:
I have no problem with paying Digium the $10 for G729 licenses,
everyone has to make money. It's the administration of the licenses
that sucks. I experiment with
We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that. That was a bit of money kissed
goodbye.
Regards,
Sahil Gupta
VoiceValley
On Sat, 3 Jun 2006, Chris Mason (Lists) wrote:
I have no problem with paying Digium the $10 for G729 licenses,
thanks William # solved my problem
/Salaque
On 6/1/06, William Piper [EMAIL PROTECTED] wrote:
That's an issue with your IP phone. Check your configuration. I believe
most phones call that digit timeout or something like that... it should be
set to about 3-4 seconds.
You can also try
On Sat, 2006-06-03 at 04:01 -0400, Chris Mason (Lists) wrote:
I have no problem with paying Digium the $10 for G729 licenses, everyone
has to make money. It's the administration of the licenses that sucks. I
experiment with different hardware a lot, and make up demo machines to
install for
Hi,
I just migrated my Asterisk installation from 1.2.1 to another server with
1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory.
All is OK but now I'm not able to see MWI indication for new messages on all
my Grandstream GXP2000 phones
make an 'lsmod' and look for any old ISDN architecture modules such as hisax
or isdn etc. There shall be no other modules loaded then hfcmulti and the
misdn stuff. You don't need CAPI, maybe this is even the clue to your not
working S0 card.
Beronet provides all you need in order to get a
Oh sorry, chan_misdn supports CAPI. The other question is wether mISDN itself
provides CAPI support..
hi
i am experiencing some problems with the configuration of an BN8S0 Beronet
card. I've downloaded last CVS of mISDN ans mISDNuser, i patch the kernel
2.6.16.18 and the enabled the
Mimmus wrote:
Hi,
I just migrated my Asterisk installation from 1.2.1 to another server with
1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory.
All is OK but now I'm not able to see MWI indication for new messages on all
my
So I took a chance with an X100P knock-off on eBay. I'm running Asterisk +
FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel
2.6.16.16. Everything has been fine up until now.
I compile the 1.2.5 Zaptel drivers without a problem, get the udev
configuration in, modprobe
On 2 Jun 2006, at 21:42, Douglas Garstang wrote:
Has anyone got any neat solutions for Asterisk .conf file revision
control?
We have multiple Asterisk boxes here, that we'd like to maintain a
_mostly_ common set of conf files on. They aren't all the same
though. There's subtle
Hi,
On 6/1/06, Mike Clark [EMAIL PROTECTED] wrote:
We have a location with around 50 Polycom phones. Asterisk version is
1.2.1 We have implemented paging through the Polycoms, which works
great. We are now trying to get FOP .26 going for the receptionist. It
seems to work fine, except that
thanks to your reply
i've also tried to use install-misdn-mqueue but:
[EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init scan
[OK] found the following devices:
card=1,0x8
[ii] run /etc/init.d/misdn-init config to store this information to
/etc/misdn-init.conf
as you can
- Bruno de Assumpção Loureiro [EMAIL PROTECTED] wrote:
This motherboard doesn't have compatibility problem in digium list
website. So, is it a compatibility problem or Digium card TE405p ??
That is a brand-new motherboard, so I doubt we've had any reports related to
it. Certainly if the
- John Fulton [EMAIL PROTECTED] wrote:
I do use featdmf in the zapata.conf.
In the zaptel.conf, it says to use the em setting for Feature Group
D, there is no featd setting in there.
Right, sorry for the confusion. It sounds like you have both files configured
properly.
--
Kevin P.
- Chris Mason (Lists) [EMAIL PROTECTED] wrote:
licenses on them, usually $100 each time, and when I install the real
hardware for the client, I can't transfer the licenses. If I scrap
that
Our support department is very accomodating when it comes to handling licensing
issues like
- Sahil Gupta [EMAIL PROTECTED] wrote:
We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that. That was a bit of money kissed
goodbye.
Unless you had been clearly abusing the key licensing system, our support
department will never refuse
- Michiel van Baak [EMAIL PROTECTED] wrote:
Then the svn automerge thingie Kevin wrote for the asterisk
svn tree is automerging changes to the 'common' tree to all
the server trees.
Glad to see someone else is making use of it too :-)
--
Kevin P. Fleming
Senior Software Engineer
Do you still have the precompiled kernel installed? Try to boot to it. Install
kernel sources and try to rerun the install-misdn-mqueue script. This is
doing the needed stuff for you. Maybe the recompilation of your kernel caused
the problem. I have installed three systems with misdn the last
thanks to your reply
using slackware the precompiled kernel is of the 2.4 series.
I've also tried to remove all modules of my 2.6 kernel, download it ,
configure it and boot it.
Then, using a new 2.6.16.18 kernel (and working) i've run the make
install of the beronet utility, but i'm still
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Friday, June 02, 2006 10:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom-Asterisk hints/presence
How do you
Here is a link that can give you the US list:
http://www.bennetyee.org/ucsd-pages/area.html
On 6/2/06, voiplist [EMAIL PROTECTED] wrote:
Is there a list somewhere or a way to find the following:1- All non US 48 area codes which can be dialed as 1+10
2- All strange area codes which are used for
Hi,
I am initiating a SIP call
from Asterisk. After about 10 minutes, I loose audio in both directions but
the call seem to stay up. Can someone please help me understand what is
happening here. Been struggling on this for a while now. This one is
preventing me from fully enjoying my
Hi,
Matthias Fechner wrote:
[portunity-in]
type=user
context=incoming-portunity
permit=82.139.223.1/255.255.255.255
now I have the next problem.
I can connect an iax phone and a sip phone to my asterisk.
The problem is with incoming phone calls.
If I use xlite everything is working perfectly
Well, I'm glad my mis-step benefited somebody else. Cost me $22 and not even sure the inbound service works yet. Their online setup docs don't match the config menus on their web site.
On 6/2/06, Ira [EMAIL PROTECTED] wrote:
At 03:03 PM 6/2/2006, you wrote:In fact, the very first words on the
Hello.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
I think that this error, is saying that its X100P is not connected in slot PCI correctly. He makes a test, he changes the X100P of slot and he sends for the list the commands lspci, dmesg, cat /proc/interrupts. I wait to have
I need a simple system with MoH, Meetme and timing using a TDM400P
with an FXO.
Any user reports?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Michael,thank´s for thisattention.
I go to test with equipment Siemens HiPath and features. I sending for you and the list an email with the results of the tests, ok? How was
zapata.conf of its asterisk with qsig? You it only changed switchtype=euroisdn, for switchtype=qsig?
Thank youfor its
Hello Josué,
yes i currently only switched switchtype in zapata.conf to the value
qsig. The only real PRI feature i've found out is the PRI_CAUSE
variable set on Hangup().
Greetings,
Michael
Josué Conti schrieb:
Michael, thank´s for this attention.
I go to test with equipment Siemens HiPath
I have a small setup with half a dozen phones and a couple of soft
phones that share a voip line and a pstn line. Right now I have it
configured to require a 9 prefix to dial to a number outside the
building. Where we are it is 10 digit dialing for even local numbers,
the local dialable area
Has anyone fed a Nortel BCM from Asterisk?
I'm interested in switching our company over, but don't want to
replace all the handsets in one fell swoop.
I imagine some of the PRI cards can emulate a switch?
I'd still like to pass CallerID into the Nortel, etc but all the
external traffic would
- Mr. Jones [EMAIL PROTECTED] wrote:
I imagine some of the PRI cards can emulate a switch?
Asterisk (and Zaptel) handles all call signaling not the cards. What that means
to you is that any Asterisk-supporting T1/E1 card can operate in PRI mode and
act as the network end as well as
At 10:06 AM 6/3/2006, you wrote:
So I was thinking that I could make it more natural to just
eliminate the requirement to dial the 9 prefix for an outside
number? Can anyone see problems with doing this?
Works perfect, jut remember to leave the code that recognizes 9 for
the people who have
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thomas Kenyon
Mimmus wrote:
Hi,
I just migrated my Asterisk installation from 1.2.1 to
another server
with 1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory.
That's right, mISDN only supports kernels up from version 2.6.9. So I see you
did have to compile a kernel yourself.
Beronet has a telephone number where they offer support. This is german one.
They also have a support mail address, just have a look at their site
http://www.beronet.com and
I've not seen an answer to this in any forum.
I make a call through Asterisk, with a VOIP phone, doesn't matter which.
The call gets made, I leave a voicemail, or complete the call in some
manner, and the other side hangs up. I hear a busy signal on the phone
on my end.
If I have an
Excellent. -
So I can basically make a crossover cable to my Nortel, and pass calls
to the old phones from the PTSN (via my VOIP originator ) in to it?
I guess I'm off to look for sample configs.
Thx
Brian
On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Mr. Jones [EMAIL
On 3 Jun 2006, at 16:11, Matthias Fechner wrote:
Hi,
Matthias Fechner wrote:
[portunity-in]
type=user
context=incoming-portunity
permit=82.139.223.1/255.255.255.255
now I have the next problem.
I can connect an iax phone and a sip phone to my asterisk.
The problem is with incoming phone
- Mr. Jones [EMAIL PROTECTED] wrote:
So I can basically make a crossover cable to my Nortel, and pass
calls
to the old phones from the PTSN (via my VOIP originator ) in to it?
Exactly. Many examples of this on the voip-info wiki.
--
Kevin P. Fleming
Senior Software Engineer
Digium,
- Rick Smith [EMAIL PROTECTED] wrote:
exten = 199,1,Answer()
exten = 199,2,Dial(SIP/100,20)
exten = 199,3,Hangup
why? And how to fix ? This is annoying...
This is handled entirely by your phone. Asterisk has already closed the channel
to the phone, so it
Hello:
I am configuring a TDM-400 card (the dev kit) with Trixbox
([EMAIL PROTECTED]). When I try to apply settings in FreePBX, the machine
locks up, except at the console, where I see
TDM PCI Master Abort
scrolling repeatedly down the screen.
I think this problem has been seen before, but
You may also have a look at
http://www.voip-info.org/wiki/view/Asterisk+mISDN+channels
thanks to your reply
using slackware the precompiled kernel is of the 2.4 series.
I've also tried to remove all modules of my 2.6 kernel, download it ,
configure it and boot it.
Then, using a new
Stephen Bosch wrote:
TDM PCI Master Abort
Does that motherboard support PCI v2.2?
Have you tried a different slot?
Is the MOLEX power connector plugged in?
Are the modules fully seated?
Jeremy McNamara
___
--Bandwidth and Colocation provided by
Have either of you any experience integrating Asterisk-related devices into existing phone equipment (trunks/pots lines, etc. (I'm somewhat new to legacy voip based telephony)) to ensure
specific or random outbound calls route through Asterisk vs bell company (ATT)? Thanks in advance,Dakota
I want to provide VoIP hosting service to 2-10+ non-profit organizations
we grant services too, and possibly some small businesses. The server environment we're looking at starting out on (systems previously used for Web development, so I have these at a very low cost over the next 18 months), is
I'm currently reviewing the latest release of FreePBX (formerly known as [EMAIL PROTECTED]). Do either of you know whether FreePBX is robust enough to handle
multiple clients, or have any recommendations on front-end Web interface to
manage client config provide clients access to manage their
Hi,
is a new port for Asterisk 1.2.8 for FreeBSD out?
Regarding to the changelog there some bugs fixed with iax and the
codecs. I hope with asterisk 1.2.8 my problem with IAX2 is solved.
Best regards,
Matthias
___
--Bandwidth and Colocation provided by
Jeremy McNamara wrote:
Stephen Bosch wrote:
TDM PCI Master Abort
Does that motherboard support PCI v2.2?
I don't know. I'll have to check. Is that a requirement?
Have you tried a different slot?
No, not yet. I have tried forcing IRQ assignments, but perhaps I need to
try a
Stephen Bosch wrote:
I don't know. I'll have to check. Is that a requirement?
Yes - Most absolutely.
http://www.digium.com/en/products/hardware/tdm400p.php
Jeremy McNamara
___
--Bandwidth and Colocation provided by Easynews.com --
Hi,
* Matthias Fechner [EMAIL PROTECTED] [03-06-06 22:13]:
is a new port for Asterisk 1.2.8 for FreeBSD out?
Regarding to the changelog there some bugs fixed with iax and the
codecs. I hope with asterisk 1.2.8 my problem with IAX2 is solved.
sry, mail should go to [EMAIL PROTECTED]
Best
On Jun 3, 2006, at 12:53 PM, Dakota Burns wrote:
Have either of you any experience integrating Asterisk-related devices
into existing phone equipment (trunks/pots lines, etc. (I'm somewhat
new to legacy voip based telephony)) to ensure specific or random
outbound calls route through
Hello Tim,
* Tim Panton [EMAIL PROTECTED] [03-06-06 19:12]:
You have a weird codec problem.
Try changing the iax config to limit it to ulaw and see if that helps:
[portunity-in]
type=user
context=incoming-portunity
permit=82.139.223.1/255.255.255.255
disallow=all
allow=ulaw
sry that
Just to close the thread. The problem was that I was using an old version of
the code.
If anyone has the same problem, you can download the code from here:
http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz
Good luck,
Danko
-
On Fri, 2006-06-02 at 12:12 -0400, Andrew Kohlsmith wrote:
The Intel g729 code is licensed for educational use ONLY. Commercial use is
forbidden without paying the patent holder. $10 a port won't break the bank
of any business with a shred of a hope of a chance of surviving, and you stay
Can someone tell me the size (or any other) limitations for the extensions.conf?
We have managed to keep our file pretty small thanks to AGI but we are
about to setup a bunch of call restrictions based on area and country
code.
One line per area code in the US alone adds a LOT of text to this
What I was attempting to visualize is the following case: 10 people in an organization pick-up their phones to make an outbound call. Before integrating Asterisk, all calls route through their current non-VoIP based phone provider. After integrating 1 trunk from a VoIP service provider into their
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.5 Echo Canceller: MG2
Failed to initailize DAA, giving up...
wcfxo: probe of :00:0e.0 failed with error -5
these lines means, your x100p is not initialized - therefore cannot
be used by zaptel. the problem below,
does chan_bluetooth working well now? (integrating sound and signal channels
in BT?) If yes, it's better than using a cheap GSM adapter (like 150Euro)?
ps: i have tested it in last year with nokia6310, but with no luck.
Just to close the thread. The problem was that I was using an old version
I believe that Cisco does the monitoring/recording that way. We've been
working with a company that has implemented Cisco's approach and they
are having problems with the recording due to network design (eg, high-
availability dual-everything. Port mirroring is only picking up half the
Dakota Burns wrote:
I'm currently reviewing the latest release of FreePBX (formerly known
as [EMAIL PROTECTED]). Do either of you know whether FreePBX is robust
enough to handle multiple clients, or have any recommendations on
front-end Web interface to manage client config
Actually, FreePBX
On Sat, Jun 03, 2006 at 11:15:57PM +0200, Woodoo People .pGa! wrote:
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.5 Echo Canceller: MG2
Failed to initailize DAA, giving up...
wcfxo: probe of :00:0e.0 failed with error -5
these lines means, your x100p is not
On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote:
Can someone tell me the size (or any other) limitations for the
extensions.conf?
We have managed to keep our file pretty small thanks to AGI but we are
about to setup a bunch of call restrictions based on area and country
code.
One line
So what are the smart folks doing when it comes to retricting/allowing
which area/country codes can and can't be called?
AGI? We can go AGI but we are trying to avoid yet more calls to AGI
apps for obvious reasons.
So, is it smarter to use AGI or have it in the text file?
Thanks..
On 6/3/06,
Jeremy McNamara wrote:
Stephen Bosch wrote:
I don't know. I'll have to check. Is that a requirement?
Yes - Most absolutely.
http://www.digium.com/en/products/hardware/tdm400p.php
I've confirmed that the board supports PCI 2.2.
I've also updated the BIOS on the motherboard, but
Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound
call. Before integrating Asterisk, all calls route through their
current non-VoIP based phone provider. After integrating 1 trunk
from a VoIP
I've been reading the Google searches trying to understand how to tie
together Adit 600 to Asterisk to provide 2 way service. I'm about blind
from reading.
I assume, the answer is using MGCP between the boxes. However, the examples
I found don't really explain fully enough to know how to
Bart Fisher wrote:
I assume, the answer is using MGCP between the boxes. However, the
examples
I found don't really explain fully enough to know how to modify
examples to work for me.
Adit 600 TDM to a Digium T1 card
The goal would be able to route calls to and from ADIT from the T1's
Asterisk with Digium's single span PRI works just fine with BCM. Contact me
off the list if you need details.
Thanks,
Wojtek
- Original Message -
From: Mr. Jones [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, June 03, 2006 1:08 PM
Subject: [Asterisk-Users]
Im trying to install and configure sangoma ...
every thing is OK but when type the command wanrouter
start the following error apears:
wan Driver not found.
Thanks for any help
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
Dakota,freepbx is a web application and associated core dialplan that allows you to do many things on top of asterisk by generating the dialplan customizations ontop of the base that it provides. Once you spend some time understanding it, you can usually do most things that you want within the
On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound
call. Before integrating Asterisk, all calls route through their
current non-VoIP based phone provider. After
Just be sure that if you ditch your POTS line that you have a proper
way to terminate 911 calls!
On 6/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up
Thanks to everyone for their tips and suggestions. I finally got the
card working by using the YellowDog Linux kernel from ppckernel.org.
There must have been some setting in the kernel config that made a
difference because the card suddenly started working after that, even
after a kernel
The Adit is realy simple as all it is is a bridge, so you have one
interface that is virtualy cross connected to another interface within
the Adit.
If you want to use the Adit with asterisk you can put the cards into
the Adit (usualy FXS and/or FXO cards) and then connect the Adit to
Asterisk, in
AFAIK 7000 lines of extesnsion.conf will not eat as much memory as an
AGI script half that long will.
Second, there is no reason that 1000 lines of code (IF you would be
adding one line for every possible area code in North America then it
would be around 800 lines, then give another 200 for the
I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes
It should work as is, just make usre that you have an extension
defined (or a catch all) for every DID you have with the provider so
that incoming works.
On 6/2/06, Steven Haldeman [EMAIL PROTECTED] wrote:
Hello,
I am attempting to figure out how to set up SIP trunking, between my company
and
Hello Michael, thank´s for help.
But what´s version asterisk you use? The qsig protocol supported for what version?
Best Regards
Josué
2006/6/3, Michael Konietzny [EMAIL PROTECTED]:
Hello Josué,yes i currently only switched switchtype in zapata.conf to the valueqsig. The only real PRI feature
Thank you for your response.All that I get when I dial in is all circutes are busy and when I dial out 503 errors. Here are my configs. Any ideas would be greatly appreciated. The provider is using a Tekelec 9000 Class 5 switch if that is any help.The provider sat us up two accounts one
I finally had to give up on extension 200. I tried deleting/recreating and reloading sipura and asterisk but no luck. I had to go to a different extension for the line 1. line 2 never acted up. The new ext works fine.
On 6/3/06, David K Parker [EMAIL PROTECTED] wrote:
I'm experiencing a problem
While sending calls to a SIP provider, the following warning generates:
-- Executing Dial(SIP/1000-c317,
SIP/[EMAIL PROTECTED]:5060|55|o) in new stack
-- Called [EMAIL PROTECTED]:5060
-- SIP/209.120.202.94:5060-0533 is making progress passing it to
SIP/1000-c317
--
- Erick Perez [EMAIL PROTECTED] wrote:
Jun 3 22:56:09 WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing
Marker
bit, because SSRC has changed
However the calls complete correctly.
I'm using 1.2.8 asterisk stable release.
It's a message that should not have been marked WARNING (or even
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application works the same way?
Or if i have SIP/g729 users and i create a
- Erick Perez [EMAIL PROTECTED] wrote:
Or if i have SIP/g729 users and i create a conference with other
users
also at g729 asterisk will not transcode (when using app_conference)?
It is not possible to mix conference audio together without converting it to an
uncompressed form first.
On Sat, 2006-06-03 at 23:20 -0500, Erick Perez wrote:
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application
I have done a lot of testing and modifications to the available
app_conference code in the last few weeks and can confirm that it is
much more efficient than using meetme in the 1.2 Asterisk tree. I have
altered app_conference to do some other things that meetme does like
entry/exit sounds and
89 matches
Mail list logo
