On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote:
does your python script generate the binary format grandstream files or do
you still need to use their closed-source tool?
Right now I'm still using their Java thing, but it's slow enough that
one of these days I guess I'll crack and
Thanks Tony!
You are (of course) absolutely correct.
I feel like an idiot for doing that when I know better.
Take care
Steve
In article
[EMAIL PROTECTED],
Steve Gladden [EMAIL PROTECTED] wrote:
I am starting over and now trying to compile/install /trunk
zaptel
libpri
asterisk
Hi list,
I was wondering i anybody ever tried to use asterisk on an openmosix
loadbalancing cluster.
Obviously, hw-related processes can not migrate from one system to
another, but any other pricess else should be able. Or not?
Hans
--
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73
hi, where can i change the sip domain?, i dont see any reference in
the docs,
thanks
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I need to configure / migrate Asterisk server from 0.9 to the latest
version with some upgrades. Please help!
Thank you.
Sincerely,
Naren Koka
(480) 829-0479
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On Mon, 30 Jan 2006, Phil Blundell wrote:
On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote:
does your python script generate the binary format grandstream files or do
you still need to use their closed-source tool?
Right now I'm still using their Java thing, but it's slow enough that
I don't think I received the whole thread, but I just wanted to mention that
the language selection has been added to the preferences page of PBX
Manager. As Stefan mentioned, it is normally done in Webmin, but since some
users may not be allowed to change anything outside of the module we added
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jerry Geis wrote:
I have a 7940 trying to connect to an existing running system.
tftp is configured and running normal.
(NOTE: I know there is a later SIP version but this is the one I have)
I see the phone bootup and ask for OS79XX.TXT which
On Mon, Jan 30, 2006 at 03:14:02PM +0200, Dmitry Ivanov wrote:
I have created dynamic CGI-like TFTP server so I will create config
files on-the-fly. Now we use this system (dynamic tftp server and Perl
CGI script) for country-wide Sipura 3000 configuration. BTW, if
anyone is interested I
On Mon, 2006-01-30 at 13:33 -0800, [EMAIL PROTECTED] wrote:
On Mon, 30 Jan 2006, Phil Blundell wrote:
On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote:
does your python script generate the binary format grandstream files or do
you still need to use their closed-source tool?
Juan Carlos Castro y Castro wrote:
How many TDM2400P cards can I safelly install in one PC? I'm loking
for
answers from whoever has a working scenario with * and a number of
cards
higher than one.
Depends on the specs of the server. For example, a quad Xeon will be
able to service
Thczv F. Thczv wrote:
I would love to run Asterisk on an old laptop, in a mostly solid state
configuration, with no HD. The laptop is slow (Pentium 233), and I
need PCMCIA support (for my network card). Are any of you aware of a
live CD that might work?
Thanks,
Dave
Take your pick:
satish Ahalawat wrote:
Hi All,
I want to setup the interconnectionm between two servers, both having
sip clients behind firewalls. I want the calls from any of the servers
to land on any of SIP clients on the other. I am looking for dial out
plans with the sample configuration files .
Hi all,
Come one come all! We're having the next Asterisk evening at the
Fujitsu Centre for Excellence! This is Fuji's state of the art
show-off centre - they're promising lots of interesting toys to play
with. As usual, we'll be discussing developments in Asterisk land over
the past couple of
Do you have the file in the SEPmac address.cnf.xml file?
i.e.
device
loadInformation model=IP Phone 7940P0S3-05-1-00/loadInformation
/device
Make sure you have SIPmac address.cnf, RINGLIST.DAT, and dialplan.xml
too.
Thanks,
Greg
-Original Message-
From: Jerry Geis [mailto:[EMAIL
Hi there guys,
Does anyone know what this is??
Every time a mISDN channel connects to anything, I get this message on
the CLI of asterisk.
Unhandled Message: prim 281 len -22 from addr 51400101, dinfo 0 on port:
1
Thanks
Pedro Nunes
___
Any idea if there will be a Sydney one?
...Skeeve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jurgen
Sent: Tuesday, 31 January 2006 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and
Business-Oriented Asterisk
On Mon, 30 Jan 2006, trixter aka Bret McDanel wrote:
I looked and it doesn't seem anyone has cracked the checksum yet.
Depending on what you want, there is a perl script called 'gsutil' that
will configure granbdstream stuff without using tftp or the web
interface. It doesnt create cfg files,
On Mon, 30 Jan 2006, Phil Blundell wrote:
/ On Mon, 2006-01-30 at 11:51 -0800, asterisk at anime.net
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
// does your python script generate the binary format grandstream files or do
// you still need to use their closed-source tool?
I played with FXO on the HT488 a bit, but didn't have a whole
lot of luck. We had a bit of a problem with echo, but more
seriously the thing kept getting itself into a variety of
wedged states: sometimes it would lock up altogether (usually
with its button lit up), and sometimes it would
The only significant feature that the SPAs seems to be
missing compared to the HTs is the Early Dial thing (where
it sends each digit to Asterisk until it gets something other
than a 484 response back).
Has anyone ever gotten that working? I've tried it on every Granstream
device I've had
Sounds great - be there and be square.
PaulH
jurgen [EMAIL PROTECTED] wrote:
Hi all,
Come one come all! We're having the next Asterisk evening at the
Fujitsu Centre for Excellence! This is Fuji's state of the art
show-off centre - they're promising lots of interesting toys to play
Anyone successfully set up one of the polycom soundpoint ip sidecars
with asterisk to monitor and allow transfer to monitored extensions?
How does it work? Any issues?
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Damon Estep wrote:
Anyone successfully set up one of the polycom soundpoint ip sidecars
with asterisk to monitor and allow transfer to monitored extensions?
How does it work? Any issues?
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On 31/01/06, Skeeve Stevens [EMAIL PROTECTED] wrote:
Any idea if there will be a Sydney one?
There was some talk about Sydney doing something, but as far as I
know, no one has done anything about it yet.
There's a page in the wiki about Sydney, with some contact information
of the person who's
I have a digium TDM400P with 1 FXO and 1 FXS port. I have a standard analog
phone connected to the FXO port and place calls to the PTSN phone line. The
analog trunk is accessed via the standard 9 and area code (if needed) and
of course the phone number. The error is as follows. I dial
This all depends on what your existing setup is, mainly where you are
storing your sip users in the first place. No point duplicating your
user list. The Ciscos consume XML, so just parse out the list of users
via some scripting language from the DB/directory/flat-files or whatever
you're
It can be done.
1. Setup a new Vm profile on CCM with a mask of
2. Setup a CTI route point:
a.Set the directory number to a
pattern. I use *27XX
but any pattern that you can send from * is
good, ie. 88XXX
b. Set the VM profile to the newly created
profile
c. Set the line to
anyone having weird problems on latest cdrtool?
#!/usr/bin/php4
Fatal error: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in /var/www/CDRTool/SOAP/client_lib.php on line 2
always get weird error like that
Delete /usr/lib/asterisk/modules/app_md5.so and update your dialplan if you use
MD5.
It is now done in functions.
/usr/lib/asterisk/modules/app_md5.so is a leftover from your previous
installation.
[app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource:
Asterisk 1.2.4 and Zaptel 1.2.3 have been released!
This update of Asterisk includes a fix for a significant memory leak in
the expression parser that is present in all previous releases of
Asterisk 1.2. This version of Zaptel includes support for the new
generation of VPM100M echo cancellation
Does anyone know what date this memory leak was introduced and/or how to
check source code for it?
I am running a pre-1.2 CVS head version and would like to know if the
potential problem exists.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-
[EMAIL PROTECTED] On Behalf
Ram-
You are three steps ahead of where you need to
be. You need to figure out what to send before you figure out how to send
it.
Add a test extension in your
extensions_custom.conf:
exten =
3852,1,Dial(sip/easycall/19197543700,30)
dial
3852 and aee if it works. If not, try:
exten =
I have several instances where conference calls are not
being torn down appropriately. My CDR shows 3000 minute calls, which are
coming in on PRI. I know that the calls aren't really lasting
thatlong. What could be causing this? IN fact, here is what
shows now:
asterisk*CLI meetmeConf
On 01/21/06 02:02 Zoa said the following:
]
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
is there any chance of a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Sunday, January 29, 2006 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] dialing 2 channels at the same time
with
Hi all,
I'm having a really frustrating time with a bunch of BT-101 phones.
They've been trouble-free and working very well for the past several
months. A couple of days ago, some of the phones (but not all of them,
yet) have started acting very strangely. All phones are running
firmware 1.0.6.7,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Tuesday, January 31, 2006 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels at the same
time withdifferent
Are you using the same NTP server for both phones?
Are you using NTP at all?
Is jitterbuffer enabled on the asterisk server?
Not sure about SIP, but on IAX if the timestamps go haywire, you can
loose audio from one side.
hth,
c
On 1/31/06, jurgen [EMAIL PROTECTED] wrote:
Hi all,
I'm
Exten = _NXXNXX,2,Set(__ORIGCID=CALLERID(number))
exten =
_NXXNXX,2,dial(sip/${EXTEN}local/[EMAIL PROTECTED]/n,r}
[alternate1]
exten = _NXXNXX,1,macro(alternate-number|${__ORIGCID})
[macro-alternate-number]
exten = s,1,set(CALLERID(number)=${ARG1})
exten = s,2,dial(SIP/[EMAIL
On Mon, Jan 30, 2006 at 04:11:12PM +0800, Dinesh Nair wrote:
is there any chance of a FreeBSD port of idefisk ? more often than not,
most of us in *BSD-land get left out in things like this. we'd be willing
to help in the port wherever we can.
1. FreeBSD can run Linux binaries, IIRC. Have
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Monday, January 30, 2006 11:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels at the same
On 01/31/06 14:24 Tzafrir Cohen said the following:
1. FreeBSD can run Linux binaries, IIRC. Have you tried the Linux
version?
nothing beats a native version, no ?
2. stick to free software ;-)
i'll ignore this in the interest of avoiding silly my license is better
than yours type
Have just done a deployment of 45 of these puppies.
They are doing their main job quite well, but of course there are minor kinks.
A not-so-minor one is that if one attempts to plug a PC into the 2nd RJ-45 jack,
as soon as you send any reasonable amount of traffic (even casual web surfing)
On Mon, Jan 30, 2006 at 03:43:18PM +0100, [EMAIL PROTECTED] wrote:
Hi,
I have a problem with setting outgoing caller id to nothing (secret)
on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID
seems to work fine when connecting the same line to a Ericsson PBX - so
Hi
as per the list people guidence
i have downloaded the Codec and installe
my Pc is P4, but i have downloaded the P2.so file
and copied in specific directory
whe i see show translation
i could able to see 30
i have configure AAH for VOIP JET connection
when i try to make call out, its using
On Tue, 2006-01-31 at 14:33 +0800, Dinesh Nair wrote:
On 01/31/06 14:24 Tzafrir Cohen said the following:
1. FreeBSD can run Linux binaries, IIRC. Have you tried the Linux
version?
nothing beats a native version, no ?
FBSD linux stuff isnt really that different from native. There are
On Mon, 2006-01-30 at 23:46 +, Chris Bagnall wrote:
Has anyone ever gotten that working? I've tried it on every Granstream
device I've had (budgetone, HT486, GXP-2000) and it's far from reliable on
any of them. Seems that when dialling an external number, the phone accepts
the first 3
Hello,
GP I'm trying to catch channel hangup in DeadAgi script.
Googling didn't help. Channel status AGI command returns 6 - line is
up, because hangup has been requested only (and not completed).
I found the following solution. In res_agi.c source I found usage of
ast_check_hangup(chan)
ram wrote:
Hi
as per the list people guidence
i have downloaded the Codec and installe
my Pc is P4, but i have downloaded the P2.so file
and copied in specific directory
whe i see show translation
i could able to see 30
i have configure AAH for VOIP JET connection
when i try to
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