Hi All;
Which is better (to have more stable or release
versions) of zaptel, libpri and asterisk: to use cvs
or svn?
In case of using cvs, why I need to type:
export
CVSROOT=:pserver:anoncvs:[EMAIL PROTECTED]:/usr/cvsroot
In other words: what is the use of pserver, anoncvs,
... with cvs
Good day all,
we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244
with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the
office through ISDN lines and then get a possibility to join meetme
conference. It works well except when customers are using SIEMENS
bilal ghayyad wrote:
Which is better (to have more stable or release
versions) of zaptel, libpri and asterisk: to use cvs
or svn?
Use SVN if you need the latest versions.
But the released versions are available as tarballs.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566
On Tue, 27 Nov 2007 09:40:56 -0500, Matt wrote:
This is a dual NAT situation. PIX on Asterisk side, and Netgear on
phone side. HOWEVER.The Asterisk box has it's own IP but it is
being tunneled through the PIX.I guess the PIX must be messing
something up?
could you post a 'sip
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Just download the g729 module that fits your hardware at
http://downloads.digium.com/pub/telephony/codec_g729/ and follow the
README: http://downloads.digium.com/pub/telephony/codec_g729/README
PS: do a 'cat /proc/cpuinfo' to know what it your
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good morning,
I would like to find a simple PCI express card with only one FXS module,
do you know where I can find such a card?
Thanks
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (Darwin)
Comment: Using GnuPG with Mozilla -
Greetings list,
I remember a discussion many months ago which ISTR concluded that asterisk
didn't play nicely at all in multi-homed setups (e.g. SIP packets not being
sent out through the same interface they were received on, etc.).
Is this still the case, or are there versions which have
On Nov 27, 2007 10:11 PM, Darrick Hartman [EMAIL PROTECTED] wrote:
There must be something in the water (or wine) in France. Nothing on
the limesurvey site requires you to register for anything. It is very
current and updated about once a month (far from abandoned). Perhaps
someone else
What is the easiest (simplest) way to do this?
--
- Eric Smith
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
which version of the pix ?
there is some bugs in old 6.3 with sip...
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matt
Envoyé : mardi 27 novembre 2007 14:11
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Asterisk behind a
On Sun, Nov 25, 2007 at 01:10:08PM +0100, Olivier wrote:
Could you get from Siemens some kind of commitment to fully support
Alert-Info or at least, to ignore Alert-Info data in incoming INVITEs ?
Siemens responded to my initial query yesterday with a rather unhelpful:
'It is an asterisk
On Wed, 2007-11-28 at 11:07 +0100, Eric Smith wrote:
What is the easiest (simplest) way to do this?
Store the dialed number in the Asterisk DB and setup an extension to
retrieve it from the DB and dial it.
Regards,
Patrick
___
--Bandwidth and
OK, I installed LimeSurvey and made up a new form.
http://winemailserver.com/survey/limesurvey/index.php?sid=94673
The account at the esurveryspro was deleted (not by me!) so there are
no results for that.
If anyone still has the patience to do this again, please go ahead.
simultaneous calls??.. will this correctly ensure the last call
retrieved from such DB was indeed the last call received?
Patrick wrote:
On Wed, 2007-11-28 at 11:07 +0100, Eric Smith wrote:
What is the easiest (simplest) way to do this?
Store the dialed number in the Asterisk DB and
randulo wrote:
OK, I installed LimeSurvey and made up a new form.
http://winemailserver.com/survey/limesurvey/index.php?sid=94673
The account at the esurveryspro was deleted (not by me!) so there are
no results for that.
If anyone still has the patience to do this again, please go ahead.
What is the easiest (simplest) way to do this?
I do it in two steps: Save the dialled number in Asterisk DB and have a
special extension (*41) which redials it.
Here is the abstract from the dialplan where I save it:
Set(_To=${EXTEN}); // Save the original extension
How does asterisk detect the loop.
What are the criteria here.
What do I need to change in the SIP message so
that asterisk will not consider it looped??
Thanks for any help
Regards
tomasz
On Nov 23, 2007 4:03 PM, Tomasz Zieleniewski [EMAIL PROTECTED]
wrote:
hi,
I use asterisk as a gateway
On Wed, 2007-11-28 at 17:08 +0530, Benjamin Jacob wrote:
simultaneous calls??.. will this correctly ensure the last call
retrieved from such DB was indeed the last call received?
Look at the subject. He said *dialled* number, not received :)
Regards,
Patrick
randulo wrote:
OK, I installed LimeSurvey and made up a new form.
You need to add the option, for those of us that are running both 1.2
and 1.4. After this Saturday, I'll be 50/50.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Hi!
1. Use group dial like in Dial(SIP/1SIP/2) and have your monitor phones
each act as SIP/2 to SIP/6 with dedicated (!) lines that have their
ringer set to silent. You might want to adjust the Caller ID name to
prefix it with the called number like to 123: from 4567890. The SNOMs
Hi Chris,
I have a multi-homed setup, and haven't had any issues, though it's
two separate network segments. My Asterisk server has one NIC
connected to our voice network (10.0.0.x), and one to our data network
(192.168.x.x). Most of my phones are connected to the voice network,
Command module show like sip
Module Description Use
Count
app_adsiprog.soAsterisk ADSI Programming Application0
chan_sip.soSession Initiation Protocol (SIP)0
2 modules loaded
-Original
This is becoming a full time job!
It occurs to me that someone should have stepped forward immediately
with the obvious: a voice recognition powered survey:
I'm sorry, did you say Debian? Is that right?
Doug: so you'll have to wait for Saturday to reply or else say which
version you use now.
No I scanned the private IP of the server, so it has to be from inside the
network.
If * is listening on 5060, why doesn't it have LISTEN next to it? Is that
shown only for TCP connections?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent:
Tomasz Zieleniewski wrote:
How does asterisk detect the loop.
What are the criteria here.
What do I need to change in the SIP message so
that asterisk will not consider it looped??
On Nov 23, 2007 4:03 PM, Tomasz Zieleniewski [EMAIL PROTECTED]
wrote:
hi,
I use asterisk as a gateway
Greetings all-
Long story short - I find myself suddenly running a Asterisk PBX after old PBX
suddenly died. Fortunately, I had been playing with Asterisk (via Trixbox)
on a server in consideration of replacing our aged Merlin Legend - so over
the course of last weekend, I brought my testbed
Sorry,
but it seems that I have banned from list.
I can reciveve, but can not send posts.
Hi!
When I use Dial(type/identifier, timeout, A(some_file))
CDR billsec starts when announcement ends. But I have to bill from when
called party answers to phone.
How can I solve my problem?
--
Suich
randulo escreveu:
OK, I installed LimeSurvey and made up a new form.
http://winemailserver.com/survey/limesurvey/index.php?sid=94673
The account at the esurveryspro was deleted (not by me!) so there are
no results for that.
If anyone still has the patience to do this again, please go
Hi, sorry for my insistence but this is a big problem for me..my steps for
remove card are ok ?
Thanks.
--
Salvatore.
- Original Message -
From: Sasa [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, November 26, 2007 4:25 PM
Subject: [asterisk-users] Remove a
Hi, sorry for my insistence but this is a big problem for me..my
steps for remove card are ok ? Thanks.
--
Salvatore.
- Original Message -
From: Sasa [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, November 26, 2007 4:25 PM
Subject: [asterisk-users]
ST == Steve Totaro [EMAIL PROTECTED] writes:
ST Trust me on this, I have tried almost everything to get it to
ST work, the best you can hope for is one way audio in a dual NAT.
ST The answer has to do with where the packets are sent from and
ST where they seem to be sent from.
I have a Nokia
Quoting Tony Plack [EMAIL PROTECTED]:
Hi, sorry for my insistence but this is a big problem for me..my
steps for remove card are ok ? Thanks.
its not much help but I have found in general asterisk is not too
graceful about zap numbering and even starting when the cards in place
don't
Hi List,
What phones support shared line appearance? I would like a phone where we
can place calls on a line and have them picked up at another phone, but we
don't want to use call parking. I want to use this in a multi tenant
environment so I would need multiple lots. Any ideals for me?
Thanks!
I ran an nmap scan on my server and got the following result:
Starting Nmap 4.23RC3 ( http://insecure.org ) at 2007-11-28 10:29 Eastern
Standard Time
Interesting ports on ool-4b7fda52.static.optonline.net (75.127.218.82):
PORT STATE SERVICE
5060/udp open|filtered sip
So I'm pretty
If you have any questions - there's forum on www.kolmisoft.com/mor to ask
questions and get answers.
Mindaugas Kezys
http://www.kolmisoft.com
Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT
Sent: Monday,
I don't know if I understood you right, but can't that be solved with call
queues?
http://www.voip-info.org/wiki/index.php?page=Asterisk+call+queues
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf
Regards,
Ricardo Carvalho
I have found out that executing AGI thru the AMI interface fill better
my needs of control. Take a look
http://bugs.digium.com/view.php?id=11282
Ignore the bug description and read the first note entry, that might
be a better way to get things done.
- Moy
On Nov 27, 2007 10:27 PM, Benjamin
Hi, sorry but perhaps I don't have explained clearly my problem...now I have
a box voip that must be replace with another box voip but I want to do test
before remove the old voip from production.
The box voip (named 1) that now is in production have three card, two isdn
card and TDM2400P that
The VoIP load includer SER on the router, only difference I am aware of.
On Nov 28, 2007 8:44 AM, Dovid B [EMAIL PROTECTED] wrote:
- Original Message -
From: David Boyd [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Well we need a light on the phone to blink when a call is on hold, but we
want to pick it up from any phone, so its a BLF key/light tied to it. Maybe
you can intergrate that with ques, I guess I need to look into that more,
just have never heard of that being done!
Thanks for the suggestion, Ill
On Wednesday 28 November 2007 08:22:16 am Leonardo Gomes Figueira wrote:
Like the previous survey the OS question is missing a Fedora option...
Or maybe I'm the only one using it ? :)
you're not the only one
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72
- Original Message -
From: David Boyd [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 26, 2007 4:29 PM
Subject: Re: [asterisk-users] OT: Best firmware for Linksys Router thatis
SIP AWARE
On Mon,
On Nov 28, 2007 9:44 AM, Dovid B [EMAIL PROTECTED] wrote:
So do I. I set SIP to high how ever the calls are still bad. I guess I need
to read up a bit more on the firmware and how to set it up correctly.
Are the calls poor quality in both directions or on just one of the
legs of the call?
I lurk and comment a little on here and have been playing with * for a
short while.
I am interested in hearing about the pros and cons for using a database
backend to Asterisk. My current setup is simple, out of the box with
config files in /etc/asterisk and logs etc going into /var.
I notice
Ron McCarthy wrote:
What phones support shared line appearance? I would like a phone where we
can place calls on a line and have them picked up at another phone, but we
don't want to use call parking. I want to use this in a multi tenant
environment so I would need multiple lots. Any ideals
On Nov 28, 2007 3:22 PM, Leonardo Gomes Figueira
[EMAIL PROTECTED] wrote:
randulo escreveu:
I never escreveux!
Like the previous survey the OS question is missing a Fedora option...
Or maybe I'm the only one using it ? :)
Actually I think Mark Spencer does too. But he's not eligible.
Do sangoma cards use the standard Zaptel drivers? Or do they have to be
compiled externally like Rhino cards?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This
On Wed, 28 Nov 2007, Administrator TOOTAI wrote:
Good day all,
we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244
with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the
office through ISDN lines and then get a possibility to join meetme
conference. It
Asterisk 1.4 im guessing? I did not know the Snom's worked with that, Ill
have to check it out then!
Thanks!
Brad
On Nov 28, 2007 9:28 AM, Russell Bryant [EMAIL PROTECTED] wrote:
Ron McCarthy wrote:
What phones support shared line appearance? I would like a phone where
we
can place calls
On Nov 28, 2007 10:52 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
Do sangoma cards use the standard Zaptel drivers? Or do they have to be
compiled externally like Rhino cards?
Sangoma maintains a patchset that gets applied to the stock zaptel
drivers before compilation. They provide automated
Oh My Word! Let's not talk about the siparator! I just had a client who
had an aweful time with it, and I never want to hear about that wretched
product again! :)
On Nov 28, 2007 2:49 AM, Vidura Senadeera [EMAIL PROTECTED] wrote:
Hi all,
use ingate siparator. www.ingate.com
ingate will
just my $0.02.
don't interduce any other point of failure into a phone system. unless
the need outweigh the disadvantage.
On 11/28/07, Alan Lord [EMAIL PROTECTED] wrote:
I lurk and comment a little on here and have been playing with * for a
short while.
I am interested in hearing about the
And they work with Asterisk/Zaptel 1.4 ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson
Sent: Wednesday, November 28, 2007 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma Question
Jeremy Mann wrote:
And they work with Asterisk/Zaptel 1.4 ?
Yes.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
--Bandwidth and
Alan Lord wrote:
I notice a great many of the contributors here seem to use a db backend
I use it to add features that that change quite often. Black lists,
meetme conferences, caller-id lookups, DID routing, fax2email, etc. I
just make sure that nothing that I add, if it were to fail
Hello List,
We purchased a TE120P card from Digium and it works great. The only
problem is that we are still experiencing echo on some calls. I've tried
various echo cancellers (right now we are using OSLEC) and still no
luck.
My question has anyone gone from the TE120P to a Sangoma
I've been complaining about this problem recently, but nothing has been done
about it.
I'm guessing some spam filtering software has gone badly wrong. The filtering
seems to be based on the content of the message rather than the sender.
On Wed, Nov 28, 2007 at 03:02:11PM +0100, Suity Zsolt
Hi.
I have a friend who is looking to replace their old Toshiba Key system
at their office. They are afraid it is going to die soon. It is over
15 years old. They asked me for advice.
I like the idea of the Asterisk Appliance AA50 for them but I do not
have any hands on experience with the
Database on a remote machine
res_pgsql.conf:
[general]
dbhost=172.16.0.2
dbport=5432
dbname=ast_config
dbuser=pbx
dbpass=
dbsock=5432
[Nov 28 20:12:02] DEBUG[6581] res_config_pgsql.c: Postgresql RealTime
Host: 172.16.0.2
[Nov 28 20:12:02] DEBUG[6581] res_config_pgsql.c: Postgresql RealTime Port:
Hi,
after some years of abstinence from bigger Asterisk installations I was
trying to give it a start again with the asteriskv6.org project. I ran
into some issues regarding the audio quality and tried the vanilla 1.4
code, with the same results.
I have asterisk (latest 1.4 branch checkout from
Pros:
1. No need to reload Asterisk when you change settings
2. Changes are applied instantly
3. Easy to manage dialplan/users/settings
4. With properly programmed GUI you can give users some self-help services
5. No noticable overhead - dual xeon + 2gb ram does 400 simm. calls
6. You can have
Bernhard Schmidt wrote:
I have had very bad call quality as soon as Asterisk needs to encode
something into alaw/ulaw (choppy sound). I tried two different clients
Check your version of GCC. If seen posts that pointed me to resolve
this issue.
Under Mandriva 2008, they are using 4.2.2
- Original Message -
From: Bernhard Schmidt
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wed,
28 Nov 2007 14:49:24 -0400
Subject: [asterisk-users] Bad audio quality in
1.4-SVN when encoding to alaw/ulaw
Hi,
Greetings and salutations.
for all tests where
Bernhard Schmidt [EMAIL PROTECTED] wrote:
Update:
Snom360 SIP/alaw Asterisk voiceprompts (demo application)
bad
unknown
Snom360 SIP/gsm Asterisk voiceprompts (demo application)
good
If each client connected only once, subsequently made a request every
minute, and then disconnected only when finished, the load might be more
reasonable. It can be a little harder to write that kind of client
though :)
Mojo
Mojo with Horan Company, LLC wrote:
So you'd be making 100
Joshua Colp [EMAIL PROTECTED] wrote:
for all tests where Snom360 was the only client on my side you can use
ekiga and get the exact same results, so this rules out a problem with
Snom. You can use ulaw instead of alaw without changes.
The pattern here is quite clear, as soon as Asterisk
You might want the directory structure at /var/lib/asterisk as well, as
it contains the current state of the voicemail boxes and any custom
sound files that might have been added
Moj
Carlos Rojas wrote:
Hello,
Only copy the configuration files, extensions.conf, sip.conf,
On 11/28/07, Jesse Molina [EMAIL PROTECTED] wrote:
I've been complaining about this problem recently, but nothing has been done
about it.
I'm guessing some spam filtering software has gone badly wrong. The
filtering seems to be based on the content of the message rather than the
sender.
Write a application to log the information to a DB, then have all other
clients connect to the database for the status. Unless I am missing
something.
-Scott
- Original Message -
From: Mojo with Horan Company, LLC [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k).
Nortel did an upgrade which changed a bunch of things today, so I thought I'd
give it another shot. It looks like I'm much closer this time, but still no
go. Can't do calling in either direction. Anyone have any
[Nov 28 15:42:41] WARNING[4098]: chan_sip.c:4957 process_sdp: Unable
to lookup host in c= line, 'IN IP4 50045'
Anyone have this problem when using T.38 faxing... and some solution perhaps?
___
--Bandwidth and Colocation Provided by
You should not care for debug messages unless you are debugging.
core show codecs
Will show you format 8 is ALAW
- Moy
On Nov 28, 2007 2:41 PM, Robert Moskowitz [EMAIL PROTECTED] wrote:
The IAX2 channel is to IAXmodem.
The SIP extension is an ATA with a fax attached.
Nov 28 15:30:20
On Nov 28, 2007 3:49 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
[Nov 28 15:42:41] WARNING[4098]: chan_sip.c:4957 process_sdp: Unable
to lookup host in c= line, 'IN IP4 50045'
Anyone have this problem when using T.38 faxing... and some solution perhaps?
Is this from a Grandstream, by
Hi guys,
My asterisk didn't send ACK for 200 ok message just for one specific
extension.
The ATA used by this extension is used by other extensions, with same
firmware version.
Looking in wireshark, I saw that ATA sent 200ok and asterisk didn't
confirm it with ACK. The ATA did this during 20s,
On Wed, 28 Nov 2007, Mojo with Horan Company, LLC wrote:
If each client connected only once, subsequently made a request every
minute, and then disconnected only when finished, the load might be more
reasonable. It can be a little harder to write that kind of client
though :)
Anthony
On Sun, 18 Nov 2007 22:14:15 +0100, Giuseppe Barichello
[EMAIL PROTECTED] wrote:
I have successfully compiled and installed Asterisk on an Alix board
(AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian
variant).
Very nice :-)
I'd rather use a PCI card to connect * to the POTS, and
The IAX2 channel is to IAXmodem.
The SIP extension is an ATA with a fax attached.
Nov 28 15:30:20 DEBUG[2997] chan_sip.c: build_route: Contact hop:
Nov 28 15:30:20 VERBOSE[3276] logger.c: -- SIP/2201-090995f0 answered
IAX2/24729-2
Nov 28 15:30:20 DEBUG[2995] chan_iax2.c: Ooh, voice format
Hello,
I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that
the MWI indicators will never go off (The blinking red light and envelope in
the LCD).
I have tried to upgrade to 1.4.14 and all different SIP versions on the
Polycoms. I am now at 1.6.7
Here is the SIP Message
I have seen this with Polycoms, ZIP2s and occassionally with Linksys 941s,
but only intermittently. Sometimes a powercycle will clear it and
sometimes not. We've never figured out what's going on, but we think it
is something to do with NAT and the phones not exactly sticking to the
spec, but
Dear Philipp;
If I used SVN, then if later I needed to do upgrade
using the make update and make upgrade, I will be able
to do it or it is a condition that I have to be used
CVS in the beginning?
Regards
Bilal
-
Which is better (to have more stable or release
Anciso, Roy wrote:
Also I called Digium about this and their
tech support does not
recommend using their HLEC software canceller on T1 cards since it
consumes so
much CPU. I was ready to get the license keys for HLEC but when I was
transferred
to sales person they
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not attempted to use much else so far.
I've ruled out
bilal ghayyad wrote:
If I used SVN, then if later I needed to do upgrade
using the make update and make upgrade, I will be able
to do it
Don't know. svn up is what I use.
or it is a condition that I have to be used
CVS in the beginning?
http://www.asterisk.org/developers/get-source tells
I think the answer to this really depends on how regularly you make
changes to your config. Adding a database connection increases the
complexity (although it is not rocket science) and adds moving parts and
therefore has more things to go wrong. If you are using Dynamic
realtime, then you are
George Pajari wrote:
[HTML-only body]
It's funny how your own message headers tell you what went
wrong:
---cut---
X-Uniserve-Spam-Score: 2.3 23 (++)
X-Uniserve-Spam-Report: Content analysis details: (2.3 points)
pts rule name description
--
On Nov 28, 2007 11:26 AM, Bruce Komito [EMAIL PROTECTED] wrote:
I have seen this with Polycoms, ZIP2s and occassionally with Linksys 941s,
but only intermittently. Sometimes a powercycle will clear it and
sometimes not. We've never figured out what's going on, but we think it
is something
Try different music. Compressed codecs are optimized for voice. MoH
that is primarily vocals may be better than non-vocal.
Darryl Dunkin wrote:
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by
On Nov 28, 2007 8:48 AM, Mindaugas Kezys [EMAIL PROTECTED] wrote:
Pros:
1. No need to reload Asterisk when you change settings
Is reloading the text based config that dangerous? Is there a memory leak
or something?
How many times can you reload before you should restart Asterisk?
-Thermal
On Nov 27, 2007 2:56 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Nov 27, 2007 at 02:22:27AM +1100, Cameron Hissey wrote:
now that you have some background,
I am having no luck installing these two cards - i have already
confirmed they are on their own IRQ etc, and if i run
The disadvantages of using the DB are:
1. Learning curve, or getting a good UI that can be flexible when
working with the DB for AMC.
2. Many features such as hinting (BLF) do not work with realtime.
Benefits:
1. Centralized storage
2. Rapid change deployment( i.e. update sip set context =
Hi folks!
I planned to put asterisk-1.4 on my slug (one of these embedded devices).
Since i don't want to compile it natively on this slow processor, i need
a cross-compile toolchain. I tried out several ways such as manually
compiling binutils and gcc, or using scratchbox. All of these with the
Anthony Francis wrote:
2. Many features such as hinting (BLF) do not work with realtime.
That's only true if *extensions.conf* comes from a
db table.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to
Thanks... That was just what I needed.
But what about going the other way? How can I pass multiple values to a
function in func_odbc?
I can't use ARRAY as it can only be used to set variables, not read form them!
Doug.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To:
George Pajari wrote:
I've had this argument with Digium before -- their position is a crock.
As you can see from my analysis below, one can certainly run the HPEC on
a full T1 PRI span if you have a fast enough processor. Digium, however,
has made the business decision not to provide HPEC
The current Digium BRI cards need the phones to send DTMF over as
SIP-INFO.
Not sure why, but googling should help. (I think this is even covered on
the Digium site)
PaulH
On Wed, 2007-11-28 at 09:47 +0100, Administrator TOOTAI wrote:
Good day all,
we have following setup: Debian Etch 64,
I also understand your stand here Kevin - there is no way you can
restrict the software running on a server out in the wild, and no way to
make sure the software they are running will not conflict in any way.
But a single port E1 card with hardware echo cancellationpossible?
PaulH
On Wed,
On Wed, Nov 28, 2007 at 10:47:44AM -0900, Mojo with Horan Company, LLC wrote:
You might want the directory structure at /var/lib/asterisk as well, as
it contains the current state of the voicemail boxes and any custom
sound files that might have been added
Voicemail boxes are actually under
On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote:
Hi, sorry but perhaps I don't have explained clearly my problem...now I have
a box voip that must be replace with another box voip but I want to do test
before remove the old voip from production.
With later versions of Zaptel you have
I am trying to get the presence/hints/BLF working along with Realtime
SIP but I never get any busy notification. core show hints always
shows the realtime sip user as idle. I have tried setting call-limit
to various values, including 1 but nothing seems to help. I have
tried limitonpeers
On Wednesday 28 November 2007 16:55:14 Douglas Garstang wrote:
Thanks... That was just what I needed.
But what about going the other way? How can I pass multiple values to a
function in func_odbc? I can't use ARRAY as it can only be used to set
variables, not read form them!
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