Hi,
this is my 1st message, I'm writing to ask if anyone knows if a PCI32
card like the TDM400P (quad analog) or the B410P (quad BRI) is working
on a PCI-X bus, at 100MHz or 133 MHz. I'm really stuck with this, since
I found a partial yes on this mailing list but my supplier says no!
Thanks,
This Friday, February 15th, at 12 Noon EST, 9AM PST, 17:00 UTC,
Lumenvox will be joining us on the VoIP Users Conference.
This week, the last in a series about IVR, Lumenvox will be there to
discuss and field your questions on their speech recognition
solutions.
http://www.VoipUsersConference.org
On Feb 13, 2008 8:48 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
Actually, I donno it is a memory leak or not. I have a server only
running asterisk. As time goes by, the free memory shown in the top
is decreased. After I restart the asterisk, the free memory comes
I observed the same
On Wed, Feb 13, 2008 at 03:48:14PM +0800, Rilawich Ango wrote:
Actually, I donno it is a memory leak or not. I have a server only
running asterisk. As time goes by, the free memory shown in the top
is decreased. After I restart the asterisk, the free memory comes
again. That's why I wonder
Hi All
What are the differences between asterisk 1.2.4 and 1.4.6 beta
In functionality ,services and bugs.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail
On Wed, Feb 13, 2008 at 01:49:38PM +1100, Mohammad Salaque wrote:
Dear all,
Anyone can point me how to soft hangup all channels using single
command ? I am using Asterisk 1.4.15.
restart now
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
On Mon, Feb 11, 2008 at 05:25:44PM +0200, Khaled Chehab wrote:
What are the differences between asterisk 1.2.4 and 1.4.6 beta
You probably ask about Asterisk 1.4 vs. Asterisk 1.6 beta, right?
In functionality ,services
You can probably read about some of the changes in the file
That's why I didn't see anything about the REALTIME function when I went
looking - many of our production systems are still on later versions of 1.2.
Given that it wasn't made obsolete at the /beginning/ of the 1.4 cycle,
I'm hoping Digium reconsider making it obsolete in 1.6 and schedule it
Hi All
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
X
X
X
X
x
192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No
215.96.142.83(None) caac0846-cf 00101/0 unkn No
192.168.8.106(None) 94910146-46 00101/0 unkn
On Feb 10, 2008 2:01 AM, Steven [EMAIL PROTECTED] wrote:
Is anyone successfully running asterisk on an HP proliant while using
their management software, hpasm?
I have two DL360's and two TE220B's. The cards have their own IRQ's.
No matter what combination of settings I use, the cards fail
Dear List,
I have to plan an instalation of an asterisk box for over 400 extensions
(Sip and Iax2) and 4 PRI channels.
I do not know which hardware (server) should I buy to support this amount of
extensions.
Someone made a similar instalation? which hardware (server) did you use?
Which was the
On Feb 13, 2008 10:15 AM, voip crazy [EMAIL PROTECTED] wrote:
Someone made a similar instalation? which hardware (server) did you use?
Which was the processor type and the amount of memory used by the server?
You will probably get some useful info on the list but also check out
voip-info.org:
Hi all,
It is posted here:
http://bugs.digium.com/view.php?id=11976
Still waiting for the approval.
Please see the notes.
thanks,
Ganbold
On 2/12/08, Johan Wilfer [EMAIL PROTECTED] wrote:
Ganbold Tsagaankhuu wrote:
Hi all,
Sorry for cross posting.
I attached my chan_ooh323
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
X
X
X
X
x
192.168.8.106(None) 04cddc1f5a0 00101/0 unkn No
215.96.142.83(None) caac0846-cf 00101/0 unkn No
192.168.8.106(None) 94910146-46 00101/0 unkn No
On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Maybe it is related but with PRI Asterisk does not generate any tone
it sends a signal regarding your keypress. If you are using SIP phones
make sure the dtmfmode in use is RFC2833.
I have just double check and my
Dear Matt;
Special thanks for you, but I did not understand what
u mean by: Hash: SHA1?
Do u mean to type SHA1 from the putty when I am
connected remotely? I tried that and I did not find
such command, but rather I found commands like
sha1sum, sha224sum, sha256sum, ...
Can u advise what
On 2/13/08, bilal ghayyad [EMAIL PROTECTED] wrote:
Dear Matt;
Special thanks for you, but I did not understand what
u mean by: Hash: SHA1?
Do u mean to type SHA1 from the putty when I am
connected remotely? I tried that and I did not find
such command, but rather I found commands like
13 feb 2008 kl. 10.27 skrev Rob Hillis:
That's why I didn't see anything about the REALTIME function when I
went looking - many of our production systems are still on later
versions of 1.2.
Given that it wasn't made obsolete at the beginning of the 1.4
cycle, I'm hoping Digium
Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote:
My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function, but it
gives OK (9 ms) back on BUSY SIP channel.
Dear list,
I need to buy a phone which could monitor the state of the maximun number of
sip extensions about 200. It is for an attendant. I just saw Snom 370 with
keypad and Linksys 962 but they do not let me to monitor 200 extensions
states adding keypads.
Do you know any kind of phone that let
On 2/13/08, voip crazy [EMAIL PROTECTED] wrote:
Dear list,
I need to buy a phone which could monitor the state of the maximun number of
sip extensions about 200. It is for an attendant. I just saw Snom 370 with
keypad and Linksys 962 but they do not let me to monitor 200 extensions
states
The norm (if memory serves) is about 64 70 extensions per attendant. After
that, people usually split off onto multiple attendants just so the
receptionists dont kill themselves in queues.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voip crazy
Sent: 13 February
On Mon, 11 Feb 2008 00:24:14 +, Cheikhou DIAW
[EMAIL PROTECTED] wrote:
i've been googling all night looking for a tutorial that shows how to make
an asterisk standalone voicemail server , no way !
Asterisk: The Future of Telephony, Second Edition
A quick look at http://ftp.digium.com/pub/asterisk/releases/ tells me
that 1.2.4 *might not* be the latest release of software.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Khaled
Chehab
Sent: 13 February 2008 09:55
To:
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so that users see the CID name + number pop up on
their computer screen, and simultaneously ring their phones.
The following script doesn't work as planned: It waits until the
script ends before moving
13 feb 2008 kl. 13.14 skrev Gergo Csibra:
Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote:
My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function, but
Hi All;
I am facing a problem that the telephon line in Egypt
does not work with the FXO port at the digium card
(TDM22B), and I tried to play in loadzone and
defaultzone without any success, when we call to the
PBX it gives Busy signal sometimes, and othertimes it
rings without any response in
Hi Bilal
could you post the TDM configuration file (zaptel.conf and zapata.conf) and
how did you compile it
Regards Ayman Date: Wed, 13 Feb 2008 04:35:43 -0800 From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com Subject: [asterisk-users] Telephone line
signaling configuration in Egypt
On Wed, Feb 13, 2008 at 04:35:43AM -0800, bilal ghayyad wrote:
Hi All;
I am facing a problem that the telephon line in Egypt
does not work with the FXO port at the digium card
(TDM22B), and I tried to play in loadzone and
defaultzone without any success, when we call to the
PBX it gives
On 13:14, Wed 13 Feb 08, Gergo Csibra wrote:
Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote:
My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function,
Vincent,
try to use System() instead of AGI()
Diego Aguirre
Infodag - Informática
FWD#: 459696
Nikotel#: 99 21 8138-2710
EnumLookup#: +55 21 8138-2710
DUNDi-br#: 21 8138-2710
Vincent escreveu:
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so
voip crazy wrote:
Dear list,
I need to buy a phone which could monitor the state of the maximun
number of sip extensions about 200. It is for an attendant. I just saw
Snom 370 with keypad and Linksys 962 but they do not let me to monitor
200 extensions states adding keypads.
I'd
On 13:46, Wed 13 Feb 08, Vincent wrote:
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so that users see the CID name + number pop up on
their computer screen, and simultaneously ring their phones.
The following script doesn't work as planned:
On Wed, Feb 13, 2008 at 02:31:11PM +0100, randulo wrote:
On Feb 13, 2008 9:29 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Gee, I only have 7 MB free! I must reboot to free some memory! And that
Asterisk is using so much memory!
Do I detect a tiny bit of sarcasm here? Someone from Digium (or
Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates.
Peoplefone is a certified partner of
Siemenshttp://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1sdc_sid=1113876080;and
AVM/FRITZ!Box http://www.fritz-shop.ch/ . Due to our rapid growth, we are
currently
On Feb 13, 2008 9:29 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Gee, I only have 7 MB free! I must reboot to free some memory! And that
Asterisk is using so much memory!
Do I detect a tiny bit of sarcasm here? Someone from Digium (or
elsewhere) might be able to jump in and explain the asterisk
Hello
This is a fun one for the list...
Twice now, the Police have contacted us to say they have had a silent
call then hangup from our landline number to the 999 service. As a
matter of course, they follow up these calls in case someone is in
distress. Nobody here was in distress - well, no
On Wed, Feb 13, 2008 at 03:02:23PM +0100, Haan Patrick wrote:
which distribution do you use?
Maybe a Fedora 7
Debian Testing here.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com
Friends,
The following mail was sent earlier to asterisk-dev and did not cause
the amount of discussion I hoped it would.
Now that we have a way to secure signalling in IAX2 and SIP in
Asterisk svn trunk, we need to start working on
the concept of a secure call - or does it really matter?
In
On Wed, 2008-02-13 at 11:33 +0200, Khaled Chehab wrote:
I am using asterisk 1.2.4
Version 1.2.4 is really quite old (it was released in January of 2006,
so is more than 24 months old at this point), and there have been
hundreds of bugs fixed since then. I'd really suggest you try a newer
On 2/13/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Feb 13, 2008 at 03:48:14PM +0800, Rilawich Ango wrote:
Actually, I donno it is a memory leak or not. I have a server only
running asterisk. As time goes by, the free memory shown in the top
is decreased. After I restart the
So that´s why I´ve always get a red bar on home screen of the Trixbox?
Phisical memory is always at top most use, near 100% (green bar turns red on
high level of memory use), and below it there is Kernel / Application,
Buffers, Cached memory uses.
tks,
On Feb 13, 2008 12:51 PM, Atis
which distribution do you use?
Maybe a Fedora 7
greez
patrick
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Tzafrir Cohen
Gesendet: Mittwoch, 13. Februar 2008 14:46
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] restart
On Tuesday 12 February 2008 23:14:58 Alex Balashov wrote:
Rizwan Hisham wrote:
Hi all,
I am planning to implement LCR routing on my already running asterisk
server. Uptill now i have found out that asterisk has no support for
lcr, i have to do something about it myself, for example using
It might be possible to get to the emergency service via 112 or a local
number as well.
Do you have *any* calls placed at about the time of the 999 calls?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Knighton
Sent: 13
As many of you may well know, Digium has been investing a great deal of
time and effort to build the very best telephony products in the
industry. We're committed to producing the highest quality hardware and
software solutions, along with things like training and support to make
your Asterisk
[EMAIL PROTECTED] wrote:
Is it important for you to conceal that a call was made from
abc to xyz on
thus-and-such a date? Or do you merely need to conceal the
content of a
call?
I was thinking about concealing called and calling number in a generic
iax2 call, I hadn't even thinked
On Wednesday 13 February 2008 08:12:25 Phil Knighton wrote:
Thing is, I have checked both our master log, and our dialled calls log
- and nobody dialled 999! Each phone has an account code applied from
sip.conf, and we log all numbers dialled. Nobody dialled out.
Have you checked all numbers
Hi all gusy,
i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go
in busy state, if you call it get the busy tone but the phone can male any
type of call.
This is my sip.conf
[502]
language = it
username = 502
secret = password
host = dynamic
type = friend
context =
Friends,
I will be attending FOSDEM in Brussells Feb 23-24. Anyone else?
Me and Philippe Sultan (the Jabber/XMPP Asterisk developer) will be
there, so we could have a SIP/XMPP/Asterisk ad hoc meeting :-)
On Thursday, Feb 21, I will be in Utrecht, Netherlands for the free
Open Telephony
Tilghman Lesher wrote:
Uh, why not? You can do LCR quite easily in the dialplan, by using func_odbc
for each of the provider lookups, then use SORT() to get the lowest cost.
It's quite easy, and you do not need to resort to AGI.
You can do almost anything in the dial plan with enough
Does anyone have any suggestions for connecting analog DID trunks? I have
some small locations that will have 2 analog DID trunks each, the only
solution that I can see will work will be using a channel bank and T1 card,
but it will be close to $1500 to terminate these DID trunks. Was hoping
On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote:
Does anyone have any suggestions for connecting analog DID trunks?
What is an analog DID trunk?
You want to connect phones to your Asterisk? Connect to the PSTN?
I have
some small locations that will have 2 analog DID trunks
Hi VoIPCrazy,
why don't you use an ATA device such as Grandstream 486 or similar?
Giorgio Incantalupo
voip crazy wrote:
Dear list,
I need to setup asterisk to send and receibe fax. I just looking about
SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc.
The asterisk box has Digium hardware, one
Dear list,
I need to setup asterisk to send and receibe fax. I just looking about
SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc.
The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO
ports).
I just read the SpanDSP (txfax and rxfax) makes the system more unstable
that
If Asterisk does indeed use SECUREDIAL or similar as distinct from
DIAL, then DIAL should wrap SECUREDIAL for calls to a party that are
secure. This would parallel HTTP GET (or POST) which use the same
function entry for both secure and insecure connections, wrapping their
secure access
An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in
On 16:59, Wed 13 Feb 08, Johansson Olle E wrote:
Friends,
I will be attending FOSDEM in Brussells Feb 23-24. Anyone else?
I'll be there (what a suprise eh ?)
Me and Philippe Sultan (the Jabber/XMPP Asterisk developer) will be
there, so we could have a SIP/XMPP/Asterisk ad hoc meeting
On Wed, Feb 13, 2008 at 11:33:19AM -0600, Tilghman Lesher wrote:
On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote:
Tilghman Lesher wrote:
Uh, why not? You can do LCR quite easily in the dialplan, by using
func_odbc for each of the provider lookups, then use SORT() to get the
On Wednesday 13 February 2008 09:57:59 Alex Balashov wrote:
Tilghman Lesher wrote:
Uh, why not? You can do LCR quite easily in the dialplan, by using
func_odbc for each of the provider lookups, then use SORT() to get the
lowest cost. It's quite easy, and you do not need to resort to AGI.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rhino's Analog cards support analog DID. no need for all the extra
stuff You will want to get an R8FXX with fxs modules that will give
you channels in sets of 2.
ADID has not really taken off in the OS telephony market I think due
to a lack of
I want to receibe the fax via mail and send faxes via web interface and a
digital send and receibe fax list.
Voipcrazy
2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]:
Hi VoIPCrazy,
why don't you use an ATA device such as Grandstream 486 or similar?
Giorgio Incantalupo
voip crazy wrote:
Hi there,
I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN
E1s.
Basically our telco is presenting calls in order of the ISDNs on our
servers.
SERVER1=1,2,3,4
SERVER2=5,6,7,8
We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in
alarm and the
Even if * is shutdown, zaptel is still running and your ISDN channels are still
technically up. Shutting down zaptel should close the channels and put those
circuits into alarm mode.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332
- Original Message -
From: Andrew
- Original Message
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 13, 2008 9:45:34 AM
Subject: Re: [asterisk-users] LCR in Asterisk
On
Wed,
Feb
13,
2008
at
11:33:19AM
-0600,
Tilghman
Lesher
wrote:
On
Wednesday
13
I'm at this moment implementing the same as you do...
Take a look at the following links:
http://blog.evaristesys.com/?p=24
http://blogtech.oc9.com/index.php?option=com_contentview=articlecatid=4:asteriskid=77:20071121astItemid=6
http://www.voip-info.org/wiki/view/Asterisk+fax
Regards,
Ricardo
Douglas Garstang wrote:
- Original Message
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 13, 2008 9:45:34 AM
Subject: Re: [asterisk-users] LCR in Asterisk
On
Wed,
Feb
13,
2008
at
11:33:19AM
-0600,
Tilghman
On Wednesday 13 February 2008 11:45:34 Jay R. Ashworth wrote:
Having programmed in about 8 different languages over the last 25
years, I can see both points of view. And admittedly, I haven't tried
to do non-trivial things with dialplan.
That said, my view of this interaction is that
Hi list,
Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed with the
latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm
not so sure that's the case.
After setting up a
At 09:33 AM 2/13/2008, you wrote:
In the same way that a PHP programmer should not attempt write Python the
way she writes PHP, I would agree with you. However, if you're willing to
adapt to the ways the dialplan works, you can create dialplans which aren't
obfuscated at all. Tcl and Lisp are
Hey, that's cool! I wish I'd known that 6 months ago.Darren Wiebe[EMAIL PROTECTED]Wed Feb 13 2008 10:33:31 AM MST from James Finstrom to Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Analog DID-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Rhino's Analog
I would recommend you use Iaxmodem / Hylafax / Avantfax for your needs.
We use this with several customers and it works very well. This way you
do not have to patch Asterisk with spanDSP. You can set up as many
virtual fax machines as your machine will handle.
On Wed, 2008-02-13 at
Doug-
Please fix your email client. One line per word in quoting is a little
excessive. Better yet, turn off HTML.
On Wednesday 13 February 2008 12:17:30 Douglas Garstang wrote:
Is that nasty little problem of no local variables in macros fixed yet?
That's a pretty big pain in the ass. You
On Wed, Feb 13, 2008 at 12:52:42PM -0600, Tilghman Lesher wrote:
On Wednesday 13 February 2008 11:45:34 Jay R. Ashworth wrote:
Having programmed in about 8 different languages over the last 25
years, I can see both points of view. And admittedly, I haven't tried
to do non-trivial things
On Wed, Feb 13, 2008 at 07:49:36PM +0100, Philipp Kempgen wrote:
Douglas Garstang wrote:
[ ... ]
do
with
a
bash
script,
as
opposed
to
Perl,
Python,
or
any
toolkits
or
frameworks.
Could you fix your e-mail client please?
I dunno; his message comes
Tilghman Lesher wrote:
Like any other language, you certainly can write in an obfuscated way, and
the dialplan does not discourage it. That said, you certainly can write in a
modularized way. I would guess that you simply aren't familiar with the
dialplan enough to make those decisions, but
this is my 1st message, I'm writing to ask if anyone knows if a PCI32
card like the TDM400P (quad analog) or the B410P (quad BRI) is working
on a PCI-X bus, at 100MHz or 133 MHz. I'm really stuck with this,
since I found a partial yes on this mailing list but my supplier says
no!
Marco,
Just check DND if it's on on the phone or not.
What is the CLI output when you try making a phone call?
Why don't you try it with a later version of astrisk and a Phone?
On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all gusy,
i have a big problem with gxp2000 and
Has anyone tried to used VB6 to communicate with the Asterisk Manager?
If so, would you be willing to share some basic code showing your
approach to getting connected and parsing results?
I've got a Telnet control that is allowing me to connect, authenticate
and see the flow of status, etc., but
On Wed, 13 Feb 2008 10:59:38 -0200, Diego Aguirre
[EMAIL PROTECTED] wrote:
try to use System() instead of AGI()
Thanks, but no go. I get an error:
[Feb 13 21:57:55] WARNING[2138]: app_system.c:107 system_exec_helper:
Unable to execute '/tmp/netcid.py|2000|Joe'
Try adding [EMAIL PROTECTED] (or what ever your voicemail contexxt is)
I've had to add the voicemail context to get MWI to work correctly in the
past.
- Original Message -
From: Jaap Winius [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 13, 2008 12:45
Is your phone actually registered to the switch. go to the CLI and do a
'sip show peers' see if extension 502 is registered. Making an outbound
call does not prove that the phone is registered.
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On 13/02/2008, Bill Andersen [EMAIL PROTECTED] wrote:
Has anyone tried to used VB6 to communicate with the Asterisk Manager?
If so, would you be willing to share some basic code showing your
approach to getting connected and parsing results?
Bill
I wrote some very very basic stuff ages ago
Khaled Chehab wrote:
Hi All
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
*569 *active SIP channels
What phones are you using? We had a similar problem with Snom 360 phones
with firmware version 6.2.2 and asterisk 1.2, whereby channels would
When I first set up asterisk, I had similar, fortunately not with the old
bill!
It basically boiled down to not enough delay between seizing the line and
dialing the digits, for example the following would have dialled the coppers
012*99 9*12345, as 012 would have been missed.
I'm guessing this
I need to setup asterisk to send and receibe fax. I just looking about
SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc.
The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO
ports).
We use (at multiple sites, mostly BRI) iaxmodem and hylafax. Extra
bonus: you get all the cool
On Wed, 13 Feb 2008 14:25:52 +0100, Michiel van Baak
[EMAIL PROTECTED] wrote:
If you want it to detach the program from it's parent you
need the double fork yes.
Thanks for the confirmation, but when doing this, the NetCID
application no longer pops up, regardless of whether I put the NetCID
code
Johansson Olle E wrote:
So please rememner that there are a few independent regular Asterisk
developers out there that is not on the Digium payroll and still take
part in decisions about Asterisk.
Point taken.
Over a year is a long time for a warning like this, considering that
Quoting Henry Devito [EMAIL PROTECTED]:
Try adding [EMAIL PROTECTED] (or what ever your voicemail
contexxt is) I've had to add the voicemail context to get MWI
to work correctly in the past.
According to the documentation, you shouldn't have to add @context
if the context is 'default'.
I am aware there is a SIP over TCP patch. Will this ever become part of
a release, if so are there any timelines?
Thanks in advance.
___
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Is it possilble for a single context to have multiple host= something like
this
[carrier]
host=ip address1
host=ip address2
host=ip address3
type=peer
disallow=all
allow=g729
allow=ulaw
canreinvite=no
insecure=yes
qualify=yes
--
Regards,
Mark Quitoriano
http://asterisk.org.ph
Fan the flame...
Looks like it is part of the 1.6 Beta.
From the Change Log:
2008-01-18 22:04 + [r99080-99085] Russell Bryant [EMAIL PROTECTED]
* CREDITS, include/asterisk/http.h, main/tcptls.c (added),
main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
main/Makefile, main/http.c,
Marco,
You should not have any issues using a PCI card in a PCI-X slot, as
long as the card is a 3.3V PCI card. The cards that you mention above
are 3.3v compatible and you should be able to use them.
All of Digium's product line is available for 3.3v slots. Most are
universal and can
SIP over TCP is included in 1.6.
http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co
On Feb 13, 2008 5:21 PM, Razza [EMAIL PROTECTED] wrote:
I am aware there is a SIP over TCP patch. Will this ever become part of a
release, if so are there any timelines?
Thanks in advance.
What phone do you use?
Linksys ?
Vieri schreef:
--- Fons van der Beek [EMAIL PROTECTED]
wrote:
Hello all,
I am using asterisk 1.4.17 together with misdn,
once in a while:
-when a call was put on hold
-the operator tries to call a internal party for
transfering the call
-the internal
As far as I'm aware, only the Aastra 57i with three 560M modules would
come close to your requirements.
The 57i can display up to 5 extensions at one time with a further 15
being available by the use of multiple pages. The 560M modules can
display up to 20 extensions at one time with three
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but what about the 'auto-${EPOCH}-' part?
I've been trying to use ${MONITOR_EXEC_ARGS} to add
To me it sounds like you should be using the Flash Operator Panel to
monitor that many extensions. The Polycom 6xx range can monitor 42
extensions.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Thursday, 14 February 2008 10:32 AM
To:
Vincent wrote:
On Wed, 13 Feb 2008 10:59:38 -0200, Diego Aguirre
[EMAIL PROTECTED] wrote:
try to use System() instead of AGI()
Thanks, but no go. I get an error:
[Feb 13 21:57:55] WARNING[2138]: app_system.c:107 system_exec_helper:
Unable to execute '/tmp/netcid.py|2000|Joe'
The
- Original Message
From: Bill Andersen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, 13 February, 2008 8:31:01 PM
Subject: [asterisk-users] Asterisk Manager and Visual Basic
Has anyone tried to used VB6 to communicate with the Asterisk Manager?
If
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