On Thu, 2011-08-04 at 11:20 -0400, Dan Journo wrote:
Hi,
Using 1.4, I see that pickupgroup can only be between 1 and 63.
We run a hosted PBX service and need to give our client access to the
call pickup feature.
I thought that I could simply use the client's ID number for the
Hey,
I have been using asterisk on slackware and had thus come up with my own
dialplan.
I would like to import my dialplan into freepbx+asterisk since I am
switching to that...how can I create my own custom dialplan in freepbx?
Thanks
Richard Zulu
Twitter
www.twitter.com/richardzulu
Skype:
Hi again,
thanks for your answer, but it didn't solve the problem. That Dial command
returns inmediately, so I don't even have the delay.
I'll try to explain myself better. The PBX has only one FXO card, connected to
the PSTN. There is no other phones connected to the PBX nor SIP extensions.
Hi!
I'm sorry that I have misundertood your question, didn't read it
carefully enough.
So you have your asterisk and your phone conntected to the same incoming
line.
Maybe you can try with to detect an answered call with BackGroundDetect()
exten = s,1,Answer()
exten =
Completely normal operation.
You need to read and understand more basic telephony and analog lines to
understand why that won't work.
Asterisk needs to be in control, and once someone answers a phone not under
Asterisk control, or the call is abandoned there is little you can do.
Sounds like a
John is absolutly right. You should connect your phones to an FXS port.
Otherwise you can't do what you wan't.
Regards,
Carlos M Cruz
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New to Asterisk?
You should change in dahdi conf the amount of time (rings) it should wait
before answering
The dialplan doesn't handle that
-Original Message-
From: Ruben Rögels ruben.roeg...@jumping-frog.org
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 05 Aug 2011 12:36:46
To: Asterisk
Maybe you can try with to detect an answered call with BackGroundDetect()
exten = s,1,Answer()
exten = s,n,BackGroundDetect(silence/10)
exten = s,n,Voicemail(1234)
I thought about that kind of solution, however...
The problem with this is that the caller will hear silence because
hello list
My question if there is any way to display a name to the customer when i
call from my Number using asterisk 1.4
Exp: I call from my number 0520 and I want when the customer recive I
call from 0520 he sees in his phone a name “test”
Thanks and regards
--
Hi
How to send REFER with replaces from asterisk (Sending out) for
doing attended transfer.
Thanks
Nikhil
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Hello,
I use asteirsk 1.6 but i think you can set the callerid variable en asterisk
1.4 to.
CALLERID(num)=test
before de dial application.
Regards
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We are having several issues with call parking in Asterisk 1.8.5.
First, when a call is parked it is announcing the park location to the
caller rather than the callee. We also are experiencing an issue
whereby if you attempt to retrieve a parked call when a new call is
incoming the new caller and
On Fri, Aug 05, 2011 at 10:59:03AM +0200, Jorge Barreiro wrote:
What I try to do is that, when there is an incoming call from the ouside, if
someone answers on a phone, then the PBX won't answer.
I have a couple of VoIP phones fed through Asterisk, as well as analogue
phones linked directly to
Hi,
thanks for your time!
O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
Completely normal operation.
You need to read and understand more basic telephony and analog lines to
understand why that won't work.
I definitely have a lot to learn yet.
Asterisk needs to be in control, and
We are having several issues with call parking in Asterisk 1.8.5.
First, when a call is parked it is announcing the park location to the
caller rather than the callee. We also are experiencing an issue
whereby if you attempt to retrieve a parked call when a new call is
incoming the new caller and
Hi,
I am using goautodial, I am using 20channels telcom PRI line and in my
server DIgium TE120 PRI card which is for 31 channel. with this
configuration
I am able to call from server . but problem whenever i restarted the server
that time is Asterisk is stop then I am not able to call outside.
On Friday 05 Aug 2011, salaheddine elharit wrote:
My question if there is any way to display a name to the customer when i
call from my Number using asterisk 1.4
Exp: I call from my number 0520 and I want when the customer recive I
call from 0520 he sees in his phone a name “test”
snip
I'll try to explain myself better. The PBX has only one FXO card, connected
to the PSTN. There is no other phones connected to the PBX nor SIP
extensions.
There are analog phones connected to the same PSTN.
What I try to do is that, when there is an incoming call from the ouside, if
On 08/04/2011 01:45 PM, Baybal Ni wrote:
I see 401, but asterisk has my proxy in its trunk list. Can this be
caused by anything else?
Some sort of failure to match the proxy to the sip.conf peer.
Is there any way to do it without using path extension?
We do it by having the proxy rewrite
Don Kelly wrote:
There are analog phones connected to the same PSTN.
And that is why what you want to do won't work. To have it do what
you'd like, you'd need to have Asterisk as the only one receiving the call.
Then your other analog phones would need to be on something like a ATAs
that
Hi
Today I upgraded my test server from 1.8.3.2 to 1.8.5 with all the
config files staying the same.
I have the line
eventfilter=!Event: FullyBooted
in my manager.conf
but since the upgrade this event is not being filtered out.
Has anyone else noticed this or is able to replicate it?
Thanks
Hi All,
When asterisk bridges a call between 2 peers and peer-A's user puts the call
on hold, then peer-A sends a INVITE with recvonly in the SDP. Asterisk
responds to peer-A with sendonly in the SDP and asterisk sends an INVITE to
peer-B with recvonly in the SDP. Peer-B then responds with a
hmm no effect. may be i shd read asterisk book for knowing the flow and
underlying architecture first.
/Zee
On Thu, Aug 4, 2011 at 4:06 PM, Israel Gottlieb isr...@gmail.com wrote:
Set(VOLUME(TX)=10) is correct but you arent putting it in a context so
asterisk doesnt know how to deal with it
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days when I call from one extension to another of the
same PBX after I dial the number the rings sound after 20 seconds.
In the CLI log, when I debug the AGI, I see always goes good until
dialparties.agi,
The portion of the script I posted only gathers the data
Add #!/usr/local/bin/perl to the top
And
my $daymax=0;
my %hourmax;
for (my $i=0;$i24;$i++) {
$hourmax{$i}=0;
}
for (my $i=0;$i$call_max;$i++) {
if ($call_con{$i}$daymax) {
$daymax=$call_con{$i};
}
if
I am using goautodial, I am using 20channels telcom PRI line and in my
server DIgium TE120 PRI card which is for 31 channel. with this
configuration
I am able to call from server . but problem whenever i restarted the
server that time is Asterisk is stop then I am not able to call
outside.
Hi all,
http://www.voip-info.org/wiki/view/Asterisk+fax shows that it is possible to
get asterisk to automatically detect a fax. However, I am using chan_ss7
instead of chan_dahdi or chan_zap. Is it possible to detect a fax with
chan_ss7?
Thanks,
Amish
--
Hi all,
I need to wait several seconds in h extension. Since Wait
application doesn't work in h extension I must use System in the
following way:
exten = h,1,
same = n,...
same = n,System(/bin/sleep 25)
same = n,...
But when I use this System command in h extension I get the
Hello.
I would like to know if is possible to send mass sms with an php agi script
through asterisk?
For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a message
via web with php and have asterisk
On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
Hi,
thanks for your time!
O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
Completely normal operation.
You need to read and understand more basic telephony and analog lines to
understand why that won't work.
I
Well even in my example there is a mistake in the second line change the 1 to a
2
exten =_.,1,Set(VOLUME(TX)=10)
exten =_.,2,Set(VOLUME(RX)=10)
-Original Message-
From: Zeeshan Ali Shah zees...@infoshield.info
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 5 Aug 2011
This is off topic...
Asterisk will not provide you with the ability to SMS random cell phones.
Being able to transport the SMS yourself is a grewling process.. Look at
software like Kamel...
Basically you have three options:
( a ) cheat and use the email method - i.e. determine everyone's
On 08/03/2011 09:33 PM, Bruce B wrote:
Can you please elaborate on how to apply the patch?
Also, is the repository updated with the new code?
No, of course not. RPMs and DEBs are not patched, they are produced when
new releases are made. If you haven't seen a new release made that
includes
Hello.
I would like to know if is possible to send mass sms with an php agi script
through asterisk?
For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a message
via web with php and have asterisk
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:42 PM
To: asterisk
Subject: [asterisk-users] Assistance sending mass sms to cellphones
Hello.
I would
Seriously Again?
This is off topic...
Asterisk will not provide you with the ability to SMS random cell phones.
Being able to transport the SMS yourself is a grewling process.. Look at
software like Kamel...
Basically you have three options:
( a ) cheat and use the email method - i.e.
Robert.
Thanks for replying.
--- On Fri, 8/5/11, Robert Huddleston rhuddles...@gmail.com wrote:
From: Robert Huddleston rhuddles...@gmail.com
Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
When you say expensive... You are talking about pennies per SMS... Again -
if you want to cheat and go the email route - that would be free... It's
unreliable and requires some thought...
If you want more information / consulting contact me off-list.
-Original Message-
From:
O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:
On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
Hi,
thanks for your time!
O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
Completely normal operation.
You need to read and understand more basic
On 08/05/2011 05:55 AM, Nikhil wrote:
How to send REFER with replaces from asterisk (Sending out) for doing
attended transfer.
The Transfer() application can be used from the dialplan to initiate a
transfer of the channel it is executed on. There's no way to do an
'automated attended
On Fri, Aug 5, 2011 at 7:53 AM, Alejandro Cabrera Obed aco1...@gmail.comwrote:
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days when I call from one extension to another of the
same PBX after I dial the number the rings sound after 20 seconds.
In
I am using the new 1.8.5 and I just found out that Asterisk won't record
the call if the call just hangup. I did a test like this:
exten = 1009, 1, Hangup()
Then I called 1009:
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [1009@init-1005:1]
Warrem thanksa lotI'll test next monday and I'll tell you.
Regards
2011/8/5 Warren Selby wcse...@selbytech.com
On Fri, Aug 5, 2011 at 7:53 AM, Alejandro Cabrera Obed
aco1...@gmail.comwrote:
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days
Jorge Barreiro wrote:
O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:
On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
Hi,
thanks for your time!
O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
Completely normal operation.
You need to read and understand
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Barreiro
Sent: Friday, August 05, 2011 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Answering machine
Use service from sms providers like smsglobal.com, they have scripts to do
that.
On Fri, Aug 5, 2011 at 9:00 PM, asterisk-users-requ...@lists.digium.comwrote:
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide
The Asterisk Development Team announces the release of Asterisk
1.6.2.20. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.20 resolves a regression that was
introduced just
prior to the release of Asterisk
Hello all,
I am struggling with an annoying problem. I have an installation with a
small number of Grandstream GXP2010 endpoints. Each endpoint has all the
extensions programmed into the phone for BLF - for instant pickup, transfer
or speed dial.
Except for the Receptionist phone, which is
O Venres, 5 de Agosto de 2011 21:20:37 Don Kelly escribiu:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge
Barreiro Sent: Friday, August 05, 2011 12:35 PM
To: Asterisk Users Mailing List -
On 08/05/2011 09:43 AM, Christian Pinedo Zamalloa wrote:
Hi all,
I need to wait several seconds in h extension. Since Wait
application doesn't work in h extension I must use System in the
following way:
exten = h,1,
same = n,...
same = n,System(/bin/sleep 25)
same =
top posting on purpose
I neglected to say all the extensions can be picked up remotely by the
other endpoints, EXCEPT the receptionist phone x3100. When calls go to that
station, they cannot be picked up. Sorry for the necessity to post twice.
/top posting on purpose
From: Cassius Smith
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