Hello
everyone,
Can someone tell me which annex the G.729 codec
from digium is.
Asterisk seems to thing it's Annex B (with a
warning in trasnlate.c)
[codec_g729b.so] = (Annex B (floating
point) G.729/PCM16 Codec Translator) == Detected 10 licensed G.729
transcodersWARNING[8192]: File
Hi eveybody again!
I don't want to be annoying, but if nobody can help me with this, I'll have to
desist of working with SIP.I have some questions about SIP, as I wrote in
another mail. I have a SIP Gateway and I have two phones (an analog one
and a DECT one) conected to it.Also, I have two
Hi All
when I modprobe the wcfxs drive and do a cat /proc/pci, it is sharing irq with
my AGP and USB, I think this is causing the card to stop working, it would work
for a couple of days or a couple of hours but then stop, I'm a complete linux
newbie, how can I force the wxfxs driver onto
Hi!
I thought it was the SIP device too, but I have looked for avery litle comand
of this device and I can't find this Ip address, and I see that its Ip is Ok,
and I have configurated the REGISTRAR section too... I don't know what's
happening, and I don't understand that, if the IP is wrong,
I think I solved the errors I was getting with my patch,
sort of anyway.
Brief over view:
Tell all the callers their position in the queue.
When they move, tell them their new position.
I was receiving Thread xxx already blocked by xxx.
I found that if I only tell caller 4 and above (Which
On Sun, 15 Jun 2003, John Laur wrote:
I do not think it is necessarily a hardware issue, as the line-in-use
lights do not light until the wcfxo kernel module is loaded. It would be
very nice for asterisk to be able to share these lines via the PBX..
That is very interesting. I have assumed
[EMAIL PROTECTED] (John Todd) writes:
Mark has fixed the REGISTER issues to be more RFC compliant. I've
created a new thread so that those of you who got bored with the old
thread might read this new one. The feature that has just been added
was added a while ago, but now it actually seems
Hi,
Does the G.729 module support adding more licences??
From what I understand the module generates a code that unlocks it for a given number
of licences..
I would probably want to buy 2 or 3 licences to test with and then later as I needed
more add then on as needed one or two at a time..
Hiya,
Yes it does.
The only thing to be careful of, as we learnt to our mistake, was that a
single purchase gives you a single key for all and thus you cannot buy 10
licenses intending to use some on one server and some on another. I guess
this would be possible by special request though.
Simon
Hi
My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe,
the IRQ to be used with a particular module?
Robb
Quoting Emanuele Pucciarelli [EMAIL PROTECTED]:
On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote:
for a couple of days or a couple of hours
Are you using voicemail2 or voicemail?
Can you confirm that /var/spool/asterisk/vm/403/INBOX has messages
and/or
/var/spool/asterisk/voicemail/default/403/INBOX has messages?
Mark
I am using voicemail2, and I can confirm that I have messages in my
inbox.
-Derek
Hi,
Im working on a call center application where callers
input some information and get transferred to an attendant, or waits in a queue
until one is available. The operator is using a PC-based system that needs to
have access to the information previously input by the caller. I was
No, you can reinstall up to 3 times I believe.
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 16, 2003 2:11 PM
Subject: Re: [Asterisk-Users] G.729 Licencing..
What if you change the hardware?
The licenses are lost?
Thanks,
Dan
-
Just a brief progress report on the the applications and dialplan not loading:
If I don't load chan_alsa.so, by using noload=chan_alsa.so in
modules.conf, I do get the dialplan, apps, and etc. (I received a hint
offlist from someone who had problems who'd tried a different version of
this
As I understand, the key you get depend on the software hardware
installation you have.
If you change Asterisk to another computer (different hardware), then you
still can use that codec?
I have installed Asterisk on a Compaq Armada 1700 notebook (celeron/300MHz)
and it works like a charm with 6
ethernet ? this is E1, so you need a balun
you should find a 406 balun at www.patton.com
or on ebay that will translate coax to rj48C
On Tuesday 27 May 2003 3:45 pm, Roger Schreiter wrote:
Eduardo Goncalves schrieb:
...
Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could
hi
using chan_capi, I get _lots_ of hanging channels after a while. This was
first beleived to be SIP related, but I doubt it. below, 'roy' is on MGCP,
and 'fax' is just a bridged dial if someone dials in, it's re-routed to
another external number
roy
asterisk1*CLI show channels
No , the bios sets the irq.
You can try to force an irq to the slot via the bios setup menu
(ie from bios setup you can set irq 5 to pci slot 2, for ex.),
or move the card to a different pci slot.
In general, pci slot #1 shares with pci slot #5, #2 with #6, 3,4 with
onboard facilities (agp,eth
If this is through your Telco, they may have turned on the Callerid-Name field along
with your number.
I had mine turn on the Callerid-Name field for us.
-Original Message-
From: Andy Powell [mailto:[EMAIL PROTECTED]
Sent: Sunday, June 15, 2003 3:25 PM
To: [EMAIL PROTECTED]
We've just moved servers and it went fine.
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 16, 2003 2:57 PM
Subject: Re: [Asterisk-Users] G.729 Licencing..
As I understand, the key you get depend on the software hardware
installation you
Hi all,
I want to install h.323 support for *, but when I launch *
from shell command asterisk -vvvc I have the next error
screen:
[chan_oh323.so]WARNING[1024]: File loader.c, Line 226
(ast_load_resource): liboh323wrap.so: cannot open shared
object file: No such file or
Hi all,
I want to install h.323 support for *, but when I launch *
from shell command asterisk -vvvc I have the next error
screen:
[chan_oh323.so]WARNING[1024]: File loader.c, Line 226
(ast_load_resource): liboh323wrap.so: cannot open shared
object file: No such file or
Hi!
I have a new problem with my SIP device.I have done some changes and
now I receive continuosly a SIP message: 501 Not impelmented back
from the SIP Gateway. I can see that it doesn't register to Asterisk.
I have in the SIP device:
Registrar 1:UnRegisteredto:
registrar:
I am running Asterisk. I want to make my local PBX. I have Cisco ATA 186-I1. i want to
connect two analog telephone connected to ATA 186 and make them extention to dial each
other. how i can make it.
Imme
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On 16/06/2003 at 10:26 DUSTIN WILDES wrote:
If this is through your Telco, they may have turned on the Callerid-Name
field along with your number.
I had mine turn on the Callerid-Name field for us.
No, not from my teleco, this is from * via the TDM card to the DECT phones
that's why it
hi
i would like samples examples to configure with isdn4linux
i have hisax card : gazel and an ISDN(BRI) line (2 channels B and 1D)
In fist time i'll use sjphone only
Perhaps there is french people on this list who can help me to do first
steps with Asterisk
thanks
Edit /etc/asterisk/manager.conf
Hi, Any of you know where to define the user
and password for gastman.???PLEAS HELP
ME!Alvaro Parres
Il lun, 2003-06-16 alle 15:22, Robert Boardman ha scritto:
My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe,
the IRQ to be used with a particular module?
I do not know to what extent you can play with the kernel code in order
to change how IRQ's are handled.
Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues
Mark,
As far as pings - we have cases when we could ping the box on both
interfaces and there are cases when we could not (we tried 3-4 sets of
NICs and drivers). All telnets, X, ssh etc. are definitely dead.
No coredumps
Steven,
The old releases are still on the server, they just don't provide a link on
their website to access it. Here are the URLs for the openh323 code that
works with chan_h323 in Asterisk.
http://www.openh323.org/bin/pwlib_1.4.11.tar.gz
http://www.openh323.org/bin/openh323_1.11.7.tar.gz
Paulo Mannheimer wrote:
Hi,
Im working on a call center application where callers input some
information and get transferred to an attendant, or waits in a queue
until one is available. The operator is using a PC-based system that
needs to have access to the information previously input by
I'm having a problem with chan_h323 compiling for Asterisk.
RedHat 7.3
PWLIB 1.4.11 pwlib_1.4.11.tar.gz
OpenH323 1.11.7 openh323_1.11.7.tar.gz
[EMAIL PROTECTED] h323]# make clean
rm -f *.o *.so core.*
[EMAIL PROTECTED] h323]# make
cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC
you need to build pwlib and/or setup your environment properly.
See asterisk/channels/h323/README
Jeremy McNamara
[EMAIL PROTECTED] wrote:
I'm having a problem with chan_h323 compiling for Asterisk.
RedHat 7.3
PWLIB 1.4.11 pwlib_1.4.11.tar.gz
OpenH323 1.11.7 openh323_1.11.7.tar.gz
[EMAIL
I found the problem. A 'make opt' doesn't create the pwlib/lib directory
when compiling pwlib. You have to do a 'make'.
I did a 'make install' for h323 but I get a Segmentation Fault when I start
Asterisk with chan_h323.
A backtrace shows the following:
(gdb) bt
#0 0x42029241 in kill () from
On Sun, 2003-06-15 at 11:37, Dan wrote:
Hi,
There is any available GSM file player for Windows, compatible with the
Asterisk GSM format?
I receive the voicemail messages by mail as attachment and the sound is in
GSM format.
Sox can convert gsm to a wav file, also you may want to mail
Yo,
I've seen very similar Zaptel-related freezes on a wide variety of
mainboards (SMP as well as non-SMP), with X100P's as well as with an E100P.
At some point, almost always at the moment a call through one of those cards
connects or disconnects, the machine completely stops responding and
use wav49 for the format.
On Mon, 2003-06-16 at 14:44, Dan wrote:
Hi Steve,
..., also you may want to mail yourself
the msgsm format which already has a header on it.
How can I do this from Asterisk?
Thanks,
Dan
- Original Message -
From: Moshe Yudkowsky [EMAIL
I have Asterisk and Cisco ATA 186. How i can make small PBX. let me know the step and
configuration made in conf files.
Imme
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I'm afraid I have no idea what your goal is here. Do you have a
phone somewhere in this configuration? I don't see it. Please
explain what it is you are trying to do. From what I see (though
much data is missing from your explanatin) anytime you place a call,
it will result in a loop.
This patch is still not bug-free. It was causing my server to crash
without warning...
One of these days I will understand Mark's full plan and outline. It
is hard, since I am
only looking at a few of the apps that are there, and have not spent
too much time
on the whole thing.
John
On
To comply with Marks request, here is a new patch against app_meetme.c
that copies data to a localbuffer. While I feel moderately comfortable
with this patch, please review my usage of strsep, malloc, and free.
When I run make from the apps/ directory I get a error message about
strsep passing
http://www.digium.com/handbook-draft.pdf
matteo.
Il lun, 2003-06-16 alle 22:24, Imran Muneer ha scritto:
I have Asterisk and Cisco ATA 186. How i can make small PBX. let me know the step
and configuration made in conf files.
Imme
___
On Mon, Jun 16, 2003 at 12:39:20AM -0500, Asterisk wrote:
Hello!
I've been following this list for several weeks now and would like to
purchase some hardware for a VOIP/ voice-mail solution. This card appears
strikingly similar to a modem. Is it? Is there a product to bring in more
Well there are some products which may fit your bill, but will take more
of your bills:
www.voicetronix.com sells a 4 port PCI FXO at about AU$1000, a 6 port
I think their 4 port card is FXS, can it be FXO as well?
-Matt
I'm sorry, I pick a bad week to stop sniffing glue, you are
Is there a product to bring in more than one POTS line short of a full
T1? It just seems silly that the technology hasn't advanced any
further than to have a single line per card.
We are working on an FXO module for the TDM400P and hope to have it ready
in a couple of months for
I've done this, with the exact versions you state, 3 times today - every one
does the full , proper thing. I did:
cd pwlib;make clean;make opt;make install
cd ../openh323;make clean;make opt;make install
cd ../asterisk/asterisk/channels/h323;make clean;make install;make samples
works every time
I have just updated to the current CVS from CVS of 12 June and I now
receive the following error message when I start *.
Freshmaker version: 62
Freshmaker passed register test
Module 0: Initialized
Module 1: Initialized
ProSLIC on module 2 failed to powerup within 510 ms
Unable to do INITIAL
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