Thanks for the input, but I still don't seem to have any luck with the
devices locking up. I've even rebuilt a new system on new hardware and a
new xorcom device but still no good. Once the device locks up that's it
the only way to get zaptel and asterisk back up is to turn them off and
restart
You mean I'm heading to NAT issues ?
And what about Record-Route options ? Will it really help to be notified of
call endings ?
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Hi,
I have two boxes :
- A asterisk server.
- A Python Server doing CTI and call control.
If a call come on the Asterisk, a sound will be played continually
Then, If somebody want to pick up this call, he will click on a Webpage
(using the Python server) that will ask the Asterisk box to
- Lee Howard [EMAIL PROTECTED] wrote:
Andrew Nowrot wrote:
I am trying to build reliable fax solution with asterisk, iaxmodem
and
hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium
3
1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After
On Mon, Jul 09, 2007 at 04:02:06PM +1000, Nathan Dennis wrote:
Thanks for the input, but I still don't seem to have any luck with the
devices locking up.
The trace you posted before mentioned tasklets. Those are not in use in
the Astribank driver code (unless you set the optional parameter
On Sun, Jul 08, 2007 at 05:58:18PM -0400, EdPimentl wrote:
Have you also consider adding adding the uBuntu steps in addition to CentOS?
-E
Ubuntu steps, due to popular demand:
apt-get install asterisk zaptel-source
m-a a-i zaptel
Untested yet. Should work on 7.04 . Bug reports are
Hi all,
I would like to know if there is any possibility to send an event when a
call is monitored?
For both start and stop monitor.
There is no event sent on asterisk 1.2 for that monitor case. I did not
find any changes regrding that on 1.4. Am I wrong?
Is it even possible to send an event
Hello list,
I have successfully set up Asterisk, but girls from our office complain
to me that when they hit Flash to transfer a call and pick the number,
they need to wait until the call is answered, and only then they could
hangup.
On the analog PBX we had before the transfer was in
Hi all,
I would like to know if there is any possibility to send an event when a
call is monitored?
For both start and stop monitor.
There is no event sent on asterisk 1.2 for that monitor case. I did not
find any changes regrding that on 1.4. Am I wrong?
Is it even possible to send an event
Is DTLS available for Asterisk on any Linux distro?
I am most interested in Centos
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If you manage to get everything working with canreinvite=yes ( i suppose u
figure out nat issues ) then you cant play music on hold , can't record
calls , and can't do most of pbx stuff asterisk is capable of .. but dont
worry asterisk doesnt disable all this features if canreinvite=on .. like if
One of my client requested that he wants to
manually shift the dial
plan like above as he has flexiable timing sometime he finishes at 3:00pm
some
time 8pm. I can
not give him freepbx access.
How about ignoring the time element completely and just telling the client to
divert his/her
hello, all of asteriskers:
i am using tdm400P in my office. i tested that TDMF generated by asterisk is so
bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18.
most of time, the # key can not be detected correctly. Does anyone has that
problem?
please give me a hit
Or, if you can have a trigger of some type. If you have say, a database,
that stores the current night service status, then you can query that
to determine if you should send the call to the after hours steps, or to
dial into the phone. Then set up another extension that the internal
people
lizhong zhu wrote:
hello, all of asteriskers:
i am using tdm400P in my office. i tested that TDMF generated by asterisk is
so bad. the sound is very soft and quality is so bad. i am using asterisk
1.2.18. most of time, the # key can not be detected correctly. Does anyone
has that problem?
Olivier wrote:
Hi,
My setup is :
PSTN - ISTP Network --- Router - Asterisk
-- SIP Phones
Phones are located in the same location.
I'm thinking about installing new phones in other locations (small
agency, home workers), registering those phones to the
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 09, 2007 4:40 PM
Subject: Re: [asterisk-users] Very bad TDMF tone !
lizhong zhu wrote:
hello, all of
I have some clients using Enswitch (Paid solution). They are real happy with it.
- Original Message -
From: Bob Gibson
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 27, 2007 7:37 PM
Subject: [asterisk-users] OpenSer/Asterisk PBX solution
We have been working a
Daniel Gradecak wrote:
Hi all,
I would like to know if there is any possibility to send an event when a
call is monitored?
For both start and stop monitor.
There is no event sent on asterisk 1.2 for that monitor case. I did not
find any changes regrding that on 1.4. Am I wrong?
Is it
I don't see the point of the service provided by GrandCentral. Party A
calls party B through GrandCentral. Party B know party A's number and
calls party A back, now party A can call party B directly, and party A
has party B's directly number.
-Original Message-
From: [EMAIL PROTECTED]
Hi Anthony,
are you sure the monitor is started and sotoped via the dialplan ?
Anthony Francis wrote:
Daniel Gradecak wrote:
Hi all,
I would like to know if there is any possibility to send an event when a
call is monitored?
For both start and stop monitor.
There is no event sent on
Anthony Francis wrote:
There are no events generated when the monitor stops and starts, but
since you are implicitly recording in your dialplan one way or another
you can just add a userevent step before recording and after.
You can also start monitoring through the Manager API in which case
GrandCentral isn't about hiding your number, it's about reachability. Grand
Central gives you a single number that rings your home, office, cell, etc...
And provides a single voicemail box for all of those numbers. As Asterisk
users, these features do not seem very ground breaking to us, as most
John Faubion wrote:
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
I've seen several people mention it taking a few days to send messages. I've
usually seen mine in a few minutes. We'll see
On 7/9/07, Daniel Gradecak [EMAIL PROTECTED] wrote:
are you sure the monitor is started and sotoped via the dialplan ?
If you're using Monitor() or MixMonitor(), then just add a UserEvent() call
just before it in the dialplan.
If you're doing monitoring of queues, it's a bit trickier - you
Hi Stefan,
actually you probably know i am using your java-asterisk :)
Yes the best solution i found till now it was to add those events to
res_monitor.c. I wonder why it was not yet done, may be there was a reason
or nobody needed it yet.
Anyhow this would be a cool feature that others should
When i send more than one messages shortly after the other, my log
(/var/spool/asterisk/sms ) looks like this
and only two of four messages arrive.
What am i doing wrong ?
I am using an AVM B1 PCI with chan-capi and 1.4.4.
and also, when sending with smsq -x only two of the messages are
Lee Jenkins wrote:
Hi all,
I'm having an odd problem with my polycom 301. I am initiating a call
to it with AMI Originate() function:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: dropped_conf=111
The to_meetme context is
On 01:06, Mon 02 Jul 07, Hans Witvliet wrote:
On Sat, 2007-04-07 at 10:57 +0200, Michiel van Baak wrote:
Read
http://svn.digium.com/view/asterisk/team/blanchet/v6/README-IPV6.txt?view=markup
before running this code.
Before taking a plunch into the code
Marc Blanchet wrote that
- Alex Robar [EMAIL PROTECTED] wrote:
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On Mon, 9 Jul 2007, Wendell Hamilton wrote:
GrandCentral doesn't do anything you can't do with asterisk. What it
does do is put those features within reach of an average person by
providing a superb user interface for the end user, which allows them to
self-administer all of these
James FitzGibbon wrote:
On 7/9/07, *Daniel Gradecak* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
are you sure the monitor is started and sotoped via the dialplan ?
If you're using Monitor() or MixMonitor(), then just add a UserEvent()
call just before it in the dialplan.
I received your message just a few minutes after you sent it; however, it
sometimes takes 3-4 days before I see messages I post coming back to me on
the list.
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original
What do I need to do to set the outbound appearance on a call so that
it shows up as Unavailable or Private?
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Hi all,
I'm running some performance tests over my Asterisk,to simple the
test,I want to configure Asterisk to allow third party registration
and invitation,so that Asterisk would not check the to head when
challenge registartion and from head when challenge invitation,and I
can only create one
Is your incoming context using chanisavail, while your internal-dialing
context is not, and just sends the call, without checking?
Mojo
Michael Wareman wrote:
Hi,
I have (to me) an interesting problem.
There are 3 physical extensions, 11, 12 and 13. All hang off Sipura
adapters.
There
The 430's have two line appearances. I'm trying to get the second line
registered to a different extension but for some reason it's not
allowing me to do this. The first line will register fine but the second
line never seems to register no matter how I swap the device ID's and
permissions
I'm looking for an easy way to make asterisk perform as a basic
(broadcast)autodialer.
Basically all I want to do is give it a list of phone #'s and a
pre-recorded message and have it call each one and play the message or
leave it on the person's answering machine.
The people I'll be calling
Hi Arun -
I need help in configuring a auto dialer system using Asterisk. I'm holding
my customers number in MySQL want to fetch 10 numbers one time and dial if
gets connected and answered by customer wants to play a sequence of message
I've tried here is my code to place calls but in this I
[EMAIL PROTECTED] wrote:
I'm looking for an easy way to make asterisk perform as a basic
(broadcast)autodialer.
Basically all I want to do is give it a list of phone #'s and a
pre-recorded message and have it call each one and play the message or
leave it on the person's answering machine.
Hi Matt -
What do I need to do to set the outbound appearance on a call so that
it shows up as Unavailable or Private?
In most cases, I think you'd need to arrange this with your provider.
If you want to do it on a call-by-call basis (in the US), dial *67
before you dial the number. If you
Shawn,
Just call and play the message and move on. Trying to find a way to
notify a couple hundred customers that their service has been changed.
Anyone have any easy ways to do this? I already have a functioning
asterisk server with a POTS interface, etc.
Set up a dial plan
i am using tdm400P in my office. i tested that TDMF generated by asterisk
is so bad. the sound is very soft and quality is so bad. i am using
asterisk 1.2.18. most of time, the # key can not be detected correctly.
Does anyone has that problem?
please give me a hit for that problem!
Hi Arun -
using php script and Asterisk manager I'm dialing numbers and once gets
connected send to an exten in my dial plan that plays an automated message
but some time without answering even it goes to my exten. How can I handle
early media in Asterisk that is I want only when user answer
Hello Users,
I have 2x2 port T1/E1 cards for sale in Pakistan.
Cards are in warrenty and going cheap as i have purchased additional cards.
regards
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Is DTLS available for Asterisk on any Linux distro?
Nope.
I've read that the reSIProcate SIP stack has DTLS support.
- Noah
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Call files and app_amd (Answering Machine Detection) come to mind.
app_amd can take a little time to tune, but you can get it to be pretty
reliable in most cases.
See: http://www.voipinfo.org/wiki/index.php?page=Asterisk+cmd+AMD
http://www.voipinfo.org/wiki/view/Asterisk+auto-dial+out
Arun Kumar wrote:
Hi
I already tried asterisk manager but Im not able to get status for each
queue member.
thanks
That must be a problem with your configuration. I get QueueMemberStatus
on my AMI interface (1.2):
Event: QueueMemberStatus
Privilege: agent,all
Queue: support
On Wed, 2007-07-04 at 09:57 -0500, John Faubion wrote:
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
I've seen several people mention it taking a few days to send messages. I've
usually seen mine
On Sun, 2007-07-01 at 18:27 -0500, Russell Bryant wrote:
Hans Witvliet wrote:
Before taking a plunch into the code
Marc Blanchet wrote that he's making code ip-version independant.
How much of these improvements have already made it into the 1.4
branche?
None, and they never will
I recently installed 1.4.5 and I've noticed a recurrence of a problem that
I thought was solved long ago, namely a very long (2-4 seconds) delay on
meetme calls. That means with two people in the conference room, it takes
2-4 seconds for what one person says to reach the other person.
Is anyone
The Asterisk development team is proud to announce a new batch of
releases. There are new releases of Asterisk and Libpri for both the
1.2 and 1.4 series.
The development team has been working especially hard on fixing bugs in
our existing release branches. These releases are regular
Thank you for your input it is very helpful
- Original Message -
From: Dovid B
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OpenSer/Asterisk PBX solution
Date: Mon, 9 Jul 2007 17:05:57 +0300
I have some clients using Enswitch (Paid
Hans Witvliet wrote:
I intended to ask, wether it would remain for the time being a
bleeding-edge-patch, or already included into the svn-tree.
Either way, i presume that i shouldn't hold my breath while waiting for
the first 1.6 ;-)) (six-months, a year?)
As far as I know, the patch is ready
There is definitely something wrong with this list.
I have my emails sorted by date, and every day, the emails do not just
come on top, but get slotted in. Today (10 July 2007), I received about
6 emails from 29th of June, couple from 30th, up until the 5th of July,
nothing of today's, or,
Noah Miller wrote:
Is DTLS available for Asterisk on any Linux distro?
Nope.
I've read that the reSIProcate SIP stack has DTLS support.
I found out that DTLS is in openSSL 0.9.8. This is available with
Redhat/Centos 5.
So the code is there. Perhaps just configuring it to some ports
Wai Wu wrote:
Hi all,
I need the zap channels going, but got the following error. What do I
need to change in my configuration? Thnx.
chan_zap.c: In function `zap_send_keypad_facility_exec':
chan_zap.c:2309: warning: implicit declaration of function
`pri_keypad_facility'
chan_zap.c:
Hi all,
I'm running some performance tests over my Asterisk,to simplify the test,I
want to configure Asterisk to allow several clients registered with
different nubmers but using single authentication account,so that Asterisk
would not check the to from head when challenging registartion
I just installed the newly released Asterisk 1.4.7 and I cannot get music
on hold. I am using the default settings with the wav files. Here is what I
get on the cli from any sip phone:
-- Executing [EMAIL PROTECTED]:1] NoCDR(SIP/1120-084e6010, ) in new
stack
-- Executing [EMAIL
What error are you getting on the Audio Codes side ? Set verbose to 5 on the
Audio codes box and try running Syslog.
- Original Message -
From: satish patel
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 26, 2007 2:14 PM
Subject: [asterisk-users] call fail from
Are you behind NAT ? Do you have canreinvite=yes ?
- Original Message -
From: Adam KOSA [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, June 25, 2007 6:37 PM
Subject: [asterisk-users] callback and bridge problem
Hi guys,
sorry for the long e-mail, i'm only trying
Hi,
I'm curious if there is any other option beside relaxdtmf in zapata , or any
where else to tune dtmf detection on TDM400 fxo boards.
in one of our sites provider is giving us 4 analog lines out of Adtran router
and Asterisk often recognize DTMF wrong.
Obviously playing with relaxdtmf was
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