Hi,
according to the wiki the value NOANSWER for the channel variable
DIALSTATUS means:
No answer. The dial command reached its number, the number rang for too
long, then the dial timed out.
In out dialplan we grap all these events with
exten = s-NOANSWER,1,Playback(sometext)
exten =
Hi,
I want that an sjphone registered using serverA can call to an sjphone
registered using serverB and vice vers. I want to know how two asterisk server
communicate to each other. Please let me know, for that, what configuration
file I have to change.
Thanking you,
Regards,
Preeta Pandey
On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez [EMAIL PROTECTED] wrote:
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
Hi Adam,
I've been googling for half an hour, i found some sort of jingle
protocol which i'm not sure what to use for but it might be the
solution? It seems to me that my problem is sending the dtmf tones, not
receiving them, so this is really gtalk related.
You've spotted the problem,
Hello,
We have an Asterisk server receiving calls using G711 (ulaw). This
server has rerouters de calls to other server using G729 (we bought the
codecs, installed, sip show channels shows the codec properly, etc.)
Using G729, there is a click while talking. Well, more than a click it
seems
On Thu, Feb 14, 2008 at 9:52 PM, Naveen Palani [EMAIL PROTECTED] wrote:
How can i pass the arguments from my dialplan to the ruby file. Is there a
way i can do it with the agi script?
Set them as variables in your extensions.conf and use them inside your
agi scripts.
raj
Jared Smith [EMAIL PROTECTED] writes:
I've been suggesting that for about four years now (long before I ever
started working for Digium), but the core Asterisk developers tell me it
will have a very negative impact on Asterisk performance.
The only reason why it has a negative impact is
I guess we ought to add ...beyond downgrading the firmware to 2.0.2 to
that. :)
Paul Hales wrote:
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and
the BLF indicators no longer function.
Has anyone had a similar issue? And a solution?
PaulH
14 feb 2008 kl. 22.35 skrev Benny Amorsen:
Matthew J. Roth [EMAIL PROTECTED] writes:
Yes, asterisk -rx will only allow you to execute CLI commands. It
also tends to spew out a bunch of garbage that makes parsing
difficult
unless verbosity is always set to 0.
It would be very handy if
15 feb 2008 kl. 07.08 skrev Kevin Kiely:
The other day my asterisk local SIP clients got hung when my
provider had a DNS failure. All registrations went dead (even the
ones that were IP addresses) and all sip peers went offline. I know
this was know problem at one point is there any
Hi,
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The only alternative available is a fully-compatible
Openvox product...but is it really fully-compatible? Any experience
about Openvox products (card and zaptel versions, etc...)?
Thank you!
Giorgio.
Thanks guys,
On two cloned machines, on one I tried:
yum install lm_sensors-devel bzip2-devel
(ignoring newt, and these were the only ones missing)
..and it compiled ok. Then on the other I just added lm_sensors-devel
and the configure -with-net-snmp worked ok, but it didn't
hi,preeta
you have to change sip.conf in both server.
suppose,
server 1 and server 2 both are asterisk server.
you want to call from server 1 to server 2.
then,
in ser-1, sip.conf
[general]
register= user:[EMAIL PROTECTED]
[user]
type=friend
fromuser=user
username=user
secret=pass
Something I've just noticed that might persuade me to move to 1.4 ... in
iax.conf, there is a new option:
transfer=mediaonly
Does this mean that it keeps itself in the loop as far as signalling/CDR
is concerned, but lets the media stream go between the 2 endpoints?
ie.
Asterisk A -
Hi Bhrugu ,
Thanks for the reply. I will check it off.
Regards,
Preeta
-Original Message-
From: [EMAIL PROTECTED] on behalf of Bhrugu Mehta
Sent: Fri 2/15/2008 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Communication between two
Al lists wrote:
Just wondering how your experience is with HPEC,
Is it just for analog interfaces or we can use it on TE122 as well?
The HPEC can be used with any Zaptel-supported interface, but we don't
provide free licenses for people to use them with T1/E1 cards, because
the potential CPU
Giorgio Incantalupo wrote:
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The only alternative available is a fully-compatible
Openvox product...but is it really fully-compatible? Any experience
about Openvox products (card and zaptel versions,
Sangoma makes a good card.
On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Giorgio Incantalupo wrote:
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The only alternative available is a fully-compatible
Openvox product...but is it really
It is fairly easy on a fresh install since the Sangoma ./Setup install
script can create all three configuration files for you.
Thanks,
Steve Totaro
On Fri, Feb 15, 2008 at 8:11 AM, Rob Hillis [EMAIL PROTECTED] wrote:
The cards themselves are okay, but the extra level of configuration is a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I would say email Kevin what he asked. The problem with switching to a
clone company is you get what you pay for. Sticking with Digium you at
least have support. and 3 clone cards and hours of troubleshooting
later you will wish you hadn't been all
The cards themselves are okay, but the extra level of configuration is a
pain in the proverbial. Zaptel is already double-configured in both
zaptel.conf and zapata.conf (that's not a complaint - I understand the
reason for the separation) but the Sangoma cards require a /third/ level
of
Hi Kevin,
unfortunately I live in Italy and you is not so easy for us to get
electronic stuff.
Let's wait and see what happens.:)
Giorgio
Kevin P. Fleming wrote:
Giorgio Incantalupo wrote:
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The
Hi Steve,
I've already tried a Sangoma card and it behaves the same as TDM400P.
But the problem arises for example when I have to change a broken card
on an old PBX keeping the modules, that's why I need a clone card like
Openvox (Sangoma modules are different as you know) Moreover I'd like
James,
If you were replying to the original post about Openvox or specified
that is what you were referring to, maybe I would not take issue but to
reply to a suggesting to use Sangoma with what you did is absolutely
misleading. There is nothing cheap or clone about Sangoma's cards.
Paul Hales wrote:
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and
the BLF indicators no longer function.
Has anyone had a similar issue? And a solution?
PaulH
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On Thu, 2008-02-14 at 21:52 +0530, Naveen Palani wrote:
How can i pass the arguments from my dialplan to the ruby file. Is
there a way i can do it with the agi script?
Sure... simply pass your arguments to the AGI() application, and they'll
show up as if they were command-line arguments to your
On Fri, 2008-02-15 at 11:58 +, Gordon Henderson wrote:
Something I've just noticed that might persuade me to move to 1.4 ... in
iax.conf, there is a new option:
transfer=mediaonly
Does this mean that it keeps itself in the loop as far as signalling/CDR
is concerned, but lets the
On Fri, 15 Feb 2008, Jared Smith wrote:
On Fri, 2008-02-15 at 11:58 +, Gordon Henderson wrote:
Something I've just noticed that might persuade me to move to 1.4 ... in
iax.conf, there is a new option:
transfer=mediaonly
Does this mean that it keeps itself in the loop as far as
Yes the 'stop gracefully' is what effectively blocks the calls as the telco
seems to take it as we are answering the calls instead of seeing them as
busy.
I will look at implementing some sort of way of busying out all the zaptel
channels, so that we eventually busy out all 120 channels (4x E1)
Steve,
Yes I work for Rhino that is no
Secret. If you read the post I was responding to the thread not pimping my own
products. I am not sure if your a Sangoma fanboy or employee since you are
apparently offended by my response, however he wasn't asking to be sold to he
was asking about
I am using IAX2, easier to get to work trow firewalls.
//Mattias
On Fri, Feb 15, 2008 at 1:14 PM, [EMAIL PROTECTED] wrote:
Hi Bhrugu ,
Thanks for the reply. I will check it off.
Regards,
Preeta
-Original Message-
From: [EMAIL PROTECTED] on behalf of Bhrugu Mehta
Sent: Fri
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Enough Said :)
Buy Digium, or Rhino, or a Knock off but avoid the witch doctor
Steve Totaro wrote:
James,
Huh? Trying to understand your rambling reply
I just like Sangoma because they just work and have excellent
support, I have no
I actually use daemon tools
http://cr.yp.to/daemontools/daemontools-0.76.tar.gz
I like it because its log handling features, it takes the stdout of asterisk
and puts it in a log directory and automatically rotates the files.
Doug Lytle wrote:
bilal ghayyad wrote:
Any script or something
James,
Huh? Trying to understand your rambling reply
I just like Sangoma because they just work and have excellent
support, I have no affiliation with them except being a very happy
customer.
You get what you pay for right? I also think Adtran or Adit are great
products. Not sure about
Johansson Olle E wrote:
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have multiple host=
something like this
First context is something we use to describe a segment of the
dialplan. I would call this section.
[carrier]
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is possible to
do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a call.
Steve Totaro wrote:
If you were replying to the original post about Openvox or specified
that is what you were referring to, maybe I would not take issue but to
reply to a suggesting to use Sangoma with what you did is absolutely
misleading. There is nothing cheap or clone about Sangoma's
On Fri, Feb 15, 2008 at 09:05:43AM -0700, Anthony Francis wrote:
I actually use daemon tools
http://cr.yp.to/daemontools/daemontools-0.76.tar.gz
I like it because its log handling features, it takes the stdout of
asterisk and puts it in a log directory and automatically rotates the
Tim Panton wrote:
The NEW frame doesn't _have_ to contain a dialed number, the digits
can be sent later
(I forget the frametype), but later means within the encrypted
session :-)
It's the DIAL command that you are thinking of. I'm considering
implementing this, but it has one major
Jim Duda wrote:
== Spawn extension (incoming-dial, fax, 0) exited non-zero on 'Zap/4-1'
Yes, I DO think that's a little odd. It should be priority 1, shouldn't it.
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When I try to make zaptel 1.4.8, I receive the following error:
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/zaptel-1.4.8/Makefile. Fix it to use EXTRA_CFLAGS. Stop.
This is on a debian 4.0 machine running linux kernel 2.6.24.2. (gcc 4.1.2).
TIA for any help in resolving
Hi Olle,
On Thu, Feb 14, 2008 at 5:35 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have multiple host=
something like this
First context is something we use to describe a segment of the
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is
possible to do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a
There are some tdm400 cards on ebay, http://search.ebay.com/tdm400
Moj
Giorgio Incantalupo wrote:
Hi,
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The only alternative available is a fully-compatible
Openvox product...but is it really
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
*snipped
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I
am
not able to get
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is
possible to do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to
Hi all,
So I'm trying to work on this complex fax server setup, and part of it involves
connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on
using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates
HylaFax+ to handle the faxing). So far,
Joshua Kinard wrote:
So I'm trying to work on this complex fax server setup, and part of it
involves connecting my asterisk server to my Rolm CBX switch, via a T1 line.
I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in
turn, activates HylaFax+ to handle the faxing).
On Sat, Feb 16, 2008 at 12:31 AM, Faruk Kasumovic [EMAIL PROTECTED]
wrote:
Johansson Olle E wrote:
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have multiple host=
something like this
First context is something we use to describe
You got me interested in this topic so I started doing some research.
There is a discussion on the asterisk-dev list about adding true busy
support to the Zaptel module. As it currently stands, when a call comes
in on a PRI channel while asterisk is shutting down asterisk sends a
signal back
On Friday 15 February 2008 01:49:46 pm Richard Lyman wrote:
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
*snipped
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even
What about the word alternatives do you not understand. Read the
title of the thread again.
Thanks,
Steve Totaro
On Fri, Feb 15, 2008 at 11:13 AM, John Faubion [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
If you were replying to the original post about Openvox or specified
that is what
On Fri, 2008-02-15 at 14:45 -0600, Anthony Messina wrote:
that does work like a charm--it sets the accountcode, except that, for some
reason, i can't access the CDR(accountcode) value during call time.
i CAN see it in channel variables, etc. but ${CDR(accountcode)} evaluates to
-Original Message-
From: Lee Howard
So, okay, there are four calls coming in on the Zap (strange, but...)
There's definitely some kind of a timing error here. I cut my channels back
down to 1, as the Rolm isn't waiting long enough for an answer back from the
asterisk server, and it
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
Moj
Jaap Winius wrote:
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but
Of course *it would be nice if* the IAX2 authentication parameters
were also encrypted, so that there was no danger of a 3rd party
hijacking your connection and generating a bunch of extra charges.
S.
On Fri, Feb 15, 2008 at 11:31 AM, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tim Panton wrote:
On Fri, Feb 15, 2008 at 08:55:11AM +0100, Johansson Olle E wrote:
I would also like to see manager wrappers that produce data that is
easy to handle for scripts, like a wrapper that produces show channels
consise in various formats. Do we have a perl programmer on
the list?
Such a generic
Joshua Kinard wrote:
Another attached text file shows what iaxmodem is doing during all of this.
Something about adjusting skew.
[2008-02-15 17:11:12] Adjusting skew to -50.
[2008-02-15 17:11:12] Adjusting skew to -100.
[2008-02-15 17:11:12] Adjusting skew to -150.
[2008-02-15
Johansson Olle E wrote:
In the long run we're trying to move to using the manager for all
parsing by adding a lot of new manager events and actions.
If there's something missing that you only can do or information you
only can get in the CLI, please tell us.
Olle,
How does what you are
I have multiple queues in my case. Do you mean multiple queues is one
of the reason to consume memory? How to only reset the queue stats?
You will see asterisk behave its worst with multiple queues and heavy
dialplan logic. I restart my boxes with queues everynight at midnight
just to reset
Hi there,
I have a cablemodem, ARRIS brand, model tm502G. It has two FXS ports.
I was wondering if anyone has details about the correct signalling of
these FXS ports when connected to original X100p.
Tests:
fxsks on the zapata.conf and zaptel.conf files. From my cellphone I
call the ARRIS, it
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Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]:
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
No, that doesn't work. ${MONITOR_FILENAME} can influence the filenames
in the string that you can tack onto the somix sequence using
${MONITOR_EXEC_ARGS}, but not the file
How about a technical comparision. What makes the Rhino better than the
Sangoma?
On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal
experience so I strongly disagree with that part of your argument.
-Original Message-
From: James Finstrom [mailto:[EMAIL
How about a technical comparision. What makes the Rhino better than the
Sangoma?
On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal
experience so I strongly disagree with that part of your argument.
-Original Message-
From: James Finstrom [mailto:[EMAIL
No. That's how we determined it was the phone and (therefore) most
likely the firmware at fault.
After we downgraded the firmware, the phone did correctly pick up it's
hints.
Sigma Networks wrote:
Paul Hales wrote:
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30)
I told myself that I was going to stay out of this one, but since you
find this important enough to reply twice to the mailing list with the
same content, it must be worth my time to reply.
If you carefully read the thread, the person who replied from Rhino went
out of his way to NOT try to sell
You are kidding, right ???
A small user that just buys one card won't get a good support from
Digium. It'll be just a waste of time on the phone.
Practically any manufacturer gives similar support including ssh'ing
in the users box.
Right now they push the user to buy a 4 channel echo canceller
On Friday 15 February 2008 23:53:19 [EMAIL PROTECTED]
wrote:
You are kidding, right ???
A small user that just buys one card won't get a good support from
Digium. It'll be just a waste of time on the phone.
Do you have experience with this or are you just talking out of your ass?
Digium
At 11:07 PM 2/15/2008, you wrote:
Really? Which manufacturers, specifically, will allow you to call up, get
remote assistance, and help you get the card working like this?
Well, Digium did this for me when I had trouble getting something to
work right with my TDM04. Took about 5 minutes with
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