From AMI you can get uptime. If the uptime is short likely Asterisk restarted.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 19, 2012, at 10:31 AM, Alex Villacís Lasso wrote:
I have a program that connects to the Asterisk Manager Interface through port
5038
Does anyone on the list have any experience with using a Sangoma D500 card with
Asterisk to transcode G729? If you could mention pros and cons I would like to
hear opinions.
Thanks
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
Is that now permit and deny are used for. To specify the acceptable IP
address(es) the user can connect from?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 26, 2012, at 10:11 AM, bilal ghayyad wrote:
Hi All;
Is it possible to restrict the authentication
I had submitted a patch some time ago to add option s to chanspy. This would
cause chanspy to exit once the specified change was not longer there. I do not
know if it ever got into a released version as I use ABE. It was not in 1.6 but
might be in 1.8.
--
Jim Dickenson
mailto:dicken
no
no yes45
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote:
On 01/12/2012 11:57 AM, Daniel - Asterisk wrote:
The simplest answer, I purchased one additional
One good thing is now that you know what the problem is you should be able to
work with zopier support and get them to fix zopier. They have been very
responsive to a couple problems I have and I am running the free version.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
One way to deal with this is to have two queues. Give priority to the original
queue callers land in. Once answered put the call in to the second queue. They
will then be in the second queue in the order the agents answered the first
queue.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
It took 36 seconds for that number to answer when I called it and it looks like
the call hung up after 32000 ms when you dialed via asterisk.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 5, 2012, at 5:45 PM, Joseph wrote:
I have a strange problem.
I'm using
If you want to stop stuff from going to the console you can use the command
logger mute and console will not get output but log file will.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 30, 2011, at 3:11 PM, Bruce B wrote:
Hi everyone,
I am playing around
as well as the debug level. These control how
much log information is generated at all not where it is being written.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 30, 2011, at 3:24 PM, Bruce B wrote:
Okay, but I thought that the line console = is supposed
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 30, 2011, at 4:55 PM, Bruce B wrote:
One can set the verbose level as well as the debug level. These control how
much log information is generated at all not where it is being written.
What do you mean
Why not use IAX trunk instead of SIP. This would make it very easy to talk
between the two * systems.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 26, 2011, at 4:07 PM, sean darcy wrote:
On 12/26/2011 05:43 PM, Yaroslav Panych wrote:
2011/12/26 sean
I would think it would be better to set a variable for each user and then have
a single context with something like:
_NXX,1,Dial(SIP/${WhatToUse}/${EXTEN})
Or something like this.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 20, 2011, at 1:03 PM, John
You also use AMI to inject audio into the conversation using the ChanSpy
application.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote:
You can’t per se, but you can call an AGI using stream?
From: asterisk-users
packet for each leg of the call. Each leg will hear
the same audio feed offset by however long it takes the packets to be
processed. In general this is a few milliseconds and should not be a big deal.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 10
.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 17, 2011, at 12:02 PM, giovanni.v wrote:
On 17/11/2011 19.45, c.savinov...@itntelecom.com wrote:
if it is what I think it is, I remember I had a similar situation a few
years ago, and I ended up having to create
Yes. If you have two asterisk boxes running you can trunk them together and
place calls from one to to the other.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 3, 2011, at 11:36 AM, mattias wrote:
if i run let's say
1 pbx running on my main linux box
I do not know if order is important but I always deny all then permit what I
want to permit.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 11, 2011, at 1:15 PM, hussein korbani wrote:
Hello,
i am having an issue with the DENY permit thingy
${CfMC_Use_CID} to get the value.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 7, 2011, at 8:03 AM, Tobias Steen wrote:
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new
it at end of call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2011, at 10:30 PM, Sam Govind wrote:
:P I'd this very similar situation/ project Carl - and guess what. The
filename is created before the call actually hits QUEUE application so these
Queue
One way of doing something when a peer registers is to use AMI to monitor
events and when a register event occurs do what you want.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 23, 2011, at 7:08 PM, Alex Balashov wrote:
On 09/23/2011 09:59 PM, CDR wrote
My provider has always sent the SIP control info from one IP and the media
packets from another. As long as your firewall passes the data there should be
no problem. I did not have to do anything special in my configuration. This is
using ABE which is based on 1.4.
--
Jim Dickenson
and as it was not I
updated it. That fixed the problem.
This system had worked before a dahdi update was applied.
Bottom line make sure you have the most current firmware for your card.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 8, 2011, at 2:49 PM, vmed
You need to use the AMI interface an deal with the events that are give to you.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 4, 2011, at 4:36 PM, bilal ghayyad wrote:
Hi All;
We know that agents can login and logout from the phone handset. But if we
need
The argument to chanspy is a pattern and not an exact match.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 2, 2011, at 3:48 PM, steve casto wrote:
asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use
flash operator panel 2.0
(from
The way I play a sound file into a bridged call is to use chanspy w option. I
do this with an application that does AMI commands.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 17, 2011, at 10:25 AM, Tom Browning wrote:
Is there an easy way to feed an audio
I get this on my Mac:
Safari can’t open the page.
Safari can’t open the page
“https://issues.asterisk.org/jira/browse/ASTERISK-17984” because Safari can’t
establish a secure connection to the server “issues.asterisk.org”.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
If I click on the link below, without jira, Safari goes to here:
https://issues.asterisk.org/main_page.php
And yes it works.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 8, 2011, at 1:54 PM, Kevin P. Fleming wrote:
On 06/08/2011 02:27 PM, Andrew Latham
In asterisk CLI do pri show spans. The fact the card is in RED alert means
the hardware does not see the pri line connected to the card.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 11, 2011, at 6:55 PM, Nicolas Ross wrote:
Le 2011-05-09 09:31, Jim Dickenson
Make sure the firmware on the card is latest. I had a problem, not like your,
and flashing the card to the latest firmware resolved it.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 9, 2011, at 6:11 AM, Nicolas Ross wrote:
Hi !
We curently have a centos 5
On May 1, 2011, at 10:09 AM, Kaushal Shriyan wrote:
On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.com wrote:
Xorcom makes a box that connects via USB that can do failover. You connect
the box to the two system via a USB cable to each system. When the box
detects
Originate successfully queued only means that the originate action was handed
off to asterisk not that is was executed yet. There are other events, depending
on which events you are reading, that tell you the call was answered and such.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
remember to look for
firmware updates. Take a look at wiki.sangoma.com and it lets you know current
firmware versions as well as how to update if you are not running the current
version.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 25, 2011, at 4:41 PM, Edwin Lam
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 5:02 AM, Vlasis Hatzistavrou wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 6:46 AM, Vlasis Hatzistavrou wrote:
Hello Jim,
Thank you
On server B use
IAX2/iax-trunk-name/whatever-needs-to-be-dialed-on-A-to-make-call in the
Dial command.
like Dial(IAX2/sfserver1/9411212)
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 12:29 PM, Alejandro Cabrera Obed wrote:
Dear, we have
to do an agent login over their LAN.
Is there any way to do this? Can the Lync server have a SIP trunk to connect to
an Asterisk box?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
--
_
-- Bandwidth
Do you have the Sangoma wanpipe software installed?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 13, 2011, at 7:37 AM, satish patel wrote:
Try dmesg command
root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0
If you want externnotify to not fire when someone checks then put in a new
option in voicemail.conf to have it work that way. Then contribute that change
and it might be accepted.
externnotify_on_check: yes|no
or some such thing.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
the job done.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 8, 2011, at 8:57 AM, Naomi Rosenberg wrote:
Thanks. That's as I thought (feared). Dial is not an option in this case but
I have come up with a workaround involving using a reference number
)
exten = _do_meetme.,n,MeetMe(${CfMC_RoomToUse},CMpqx1)
exten = _do_meetme.,n,UserEvent(AfterMeetMe,ActionID:${CfMC_CurrentID}
Room:${CfMC_RoomToUse} ${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 28, 2011
If you want total control from AMI then point at an extension that you can set
variables to commands and arguments, call an AGI and set variables that can be
passed back to AMI via user events.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 16, 2011, at 3:03 PM
What we do is just before the call to queue we do a userevent that has the
uniqueid and the channel and any other information we care about. You can hold
on to this information and match it when you get the agentconnect event.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
option then * will
exit the chanspy app and you can loop back to the top.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 9, 2011, at 6:28 PM, Raj Mathur (राज माथुर) wrote:
Hi,
I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5
digits
that basically does an agentlogin.
Ideas as to how to best accomplish this would be appreciated.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api
You might be able to use a macro on the dial command (option M) which gets run
when the remote end answers.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 29, 2011, at 10:30 AM, Joseph Begumisa wrote:
Hi,
I have a situation where a call comes in to my
You can always place a call to an extension that sends a user event from AMI.
If there are no native AMI commands that can return what you want originate a
call to a local extension that returns a user event.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10
It should not be too hard to write some dialplan code that detects the busy,
plays a sound file asking if you want to camp-on to the called device, read an
answer and loop around checking device status and placing a call when both the
calling device and called device are free.
--
Jim Dickenson
= get_info,n,UserEvent(GetInfo,Version:ABE
DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)} CfMC:83351)
exten = get_info,n,Hangup()
exten = do_noop,1,Answer()
exten = do_noop,n,Wait(1)
exten = do_noop,n,Hangup()
You would then do what you need to do in your extensions.
--
Jim Dickenson
= sys.domain.com-1294514614.2629
CallerID= 444555
CallerIDName= Jim Dickenson
DNIDDigits= 9111222
RDNIS= (N/A)
State= Up (6)
Rings= 0
NativeFormat= 0x2 (gsm)
WriteFormat=0x2 (gsm)
ReadFormat= 0x2 (gsm
this without analyzing
the audio stream?
Are there reasons to prefer the use of PRI over SIP or SIP over PRI?
I would like people's opinions as to if one form is better than the other in
any meaningful way.
Thanks for you feed-back.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
If you set bindaddr in any conf file you will need to change the IP address
there.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 24, 2010, at 4:30 AM, Asterisk Man wrote:
Friends,
Do we need to change any Asterisk configuration files (Or any file related
If on the dial command you add option g, if the call is not answered, it will
fall through to the next statement which can be a hangup command and then it
will go to the h extension. If that does not then make the statement after the
dial command a goto h extension.
--
Jim Dickenson
I do not have log examples to provide but do have info about other issues.
There is a nocolor option in asterisk.conf that can turn off color.
logger.conf has a provision to use syslog directly.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 9, 2010, at 5:57 AM
somehow say I want to
see events with names x,y,z and forget the whole class mess. That way if you
want to see some specific events but not others you could get what you want.
I know this does not answer you question but I for sure feel your pain.
--
Jim Dickenson
mailto:dicken...@cfmc.com
Can also do asterisk -r -x 'restart now'
asterisk*CLI help restart
restart gracefully Restart Asterisk gracefully
restart now Restart Asterisk immediately
restart when convenient Restart Asterisk at empty call volume
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
What you did is what I would have done. That way the executables have their
conf file location adjusted and everything will be inside the specified
--prefix location.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 20, 2010, at 5:48 AM, Stephen Brown wrote
For sure DAHDI and libpri support is there for Asterisk 1.4.x. I am not sure
either are tied to a specific version of Asterisk as it is the chan_dahdi
module that interfaces with DADHI.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 14, 2010, at 8:38 AM, Gordon
You get into asterisk by saying asterisk -r. You then up the verbosity by
saying core set verbose 3 or some such number. You the call your number and
you should see the steps of your dialplan execute.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 13, 2010, at 7
The other thing you can do is put UserEvent() calls in your dialplan that can
have pretty much anything you want in them.
exten = s,5,UserEvent(DidQueue,${UNIQUEID} ${CHANNEL})
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 8, 2010, at 10:45 AM, Miguel Molina
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 29, 2010, at 10:20 AM, A J Stiles wrote:
On Wednesday 29 Sep 2010, Songtao Yu wrote:
Hi All,
When I tried to write my dial plan as below for my FXO port, which connects
one PSTN line:
[from-pstn]
exten =s
Did you install the header files after ./configure was run? If so redo the
./configure command and see what that does.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2010, at 3:57 PM, Danny Dias wrote:
Hello Paul,
Here is the output of the commands:
r
Do you not need to do a ./configure command before make make install? If so
issue the ./configure command again and see if that fixes the problem.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2010, at 4:32 PM, Danny Dias wrote:
Thanks Jim,
What do you
} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 20, 2010, at 5:51 AM, Jon Farmer wrote:
Hi
I have a call established and I want to play audio to just one channel
on that call. Is this possible? If so
* to stop.
https://issues.asterisk.org/view.php?id=14594
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 1, 2010, at 6:24 AM, Rushikesh wrote:
Hi list,
Im using asterisk 1.6.0.10 and have following dialplan for doing chanspy
[app-chanspy]
include = app
I had the same need which is why I submitted the patch. I think the feature
might finally be added to 1.8, it I remember correctly. I am not aware of any
other way around this.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 1, 2010, at 9:12 AM, Rushikesh wrote
= do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 9, 2010, at 5
extension to
track hang ups. We set an action token variable to a unique value for each
originate and all the user events have this token so we can tie them back to
the original originate. We turn off most AMI read message classes to cut down
on packet volume.
--
Jim Dickenson
mailto:dicken
adding option g to
the dial command and then retrieve the variable you set in the macro after the
dial step finishes. I do not use agi's much so I might be off base with all
this but it is something to look in to.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 29
Depending on what you are recording there might be two files, one for each leg
of a call, until the call ends and the files are mixed.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 29, 2010, at 7:46 AM, Motiejus Jakštys wrote:
On Thu, Jul 29, 2010 at 2:32 PM
Do you have your softphone setup to use a stun server so it can send it's
public IP address in the SIP packets? I see in the SIP debug output a 192.168
address for the RTP packets to go to which of course will not work.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
Which version of Asterisk are you running?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 2:19 AM, Zarko Zivanovic wrote:
Great, but how exactly do i find that channel - that is my question - which
command.
I am using ruby instead of agi - and i
that are available to you. ${CHANNEL} will have SIP/
voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here’s something that should be easy for RUBY pro’s.
Here is a script
you should have been able to ask more directed
questions.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 9:28 AM, Zarko Zivanovic wrote:
Jim thanks.
I will test this first thing in the morning as I am out of the office now. As
a matter of fact I
Depending on the version of Asterisk you are running you can call a macro or an
agi as option to dial. These will be called when the line is answered and you
can find the channel name of who answered.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 5
You should be able to compile the new version, stop asterisk then make install.
If you do not do make samples then your conf files will be left alone. Once you
have done make install you can the start asterisk again.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
a multi channel
dial command. Is this not what you wanted to know?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 10:40 AM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables
What do you mean now that ABE is discontinued? My company payed thousands of
dollars this year for the product and the support it provides!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 5, 2010, at 5:42 PM, Carlos Chavez wrote:
On Mon, 2010-07-05 at 19:59
Yes it gets called when the call is connected to a queue member.
In version 1.4.x you can execute an AGI instead of a sub or macro.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 7:20 AM, Jonas Kellens wrote:
Hello list,
I notice on the wiki
= AGITool_exec(agi, res, UserEvent, Event);
// sprintf(Event, Do UserEvent = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
AGITool_Destroy(agi);
return 0;
} /* main */
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 8:31 AM, Jonas
You might take a look at the SIPHEADER function which can return specific SIP
headers.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 7:36 PM, kamrun nahar bina wrote:
Dear all,
I want to retrieve the value from Contact header and from From
What OS are you running on the two systems?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 15, 2010, at 10:40 AM, Faisal Hanif wrote:
Till now I am not able to find any difference between both machines.
Can you please tell me how I can try to resolve it on OS
One thing I would do is something like
On system one:
rpm -qa | sort sys1
On system two:
rpm -qa | sort sys2
Then on either system do a diff of these two files. If you only use yum or rpm
to install and update software you can tell what is different between the two
systems.
--
Jim Dickenson
Starting with version 1.6.x there is a VERSION function that I think will give
you the version number.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 9, 2010, at 5:19 AM, Vieri wrote:
Simple enough:
How can I get Asterisk version from within my dialplan
we end up with hundreds of calls that
Asterisk thinks have been dealt with but the provider still thinks are pending
some action. Can something be done from the Asterisk side to deal with this
situation?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
user manual sure looks like my setup. I contacted
Sangoma support and they say the product does not do NAT transversal.
Has anyone found a work around this problem?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
to be executed on the
calling party's channel once they are connected to a queue member.
Newer versions allow for either an AGI or a macro.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 20, 2010, at 4:47 AM, Vasiliy G Tolstov wrote:
Hello.
Can You provide example
Use read application
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 19, 2010, at 9:42 AM, taimur hasan wrote:
Hello
I am new to Asterisk. I want to know is there any way to get DTMF input from
the user in the Dialplan.
Regards
Taimur Hasan
-THQ
The two phones belong to context phones and the two extensions are in context
internal. In context phones you need to include = internal so that context
phones knows about those extensions. Or put the two extensions in context
phones and not context internal.
--
Jim Dickenson
mailto:dicken
= 201,n,Verbose(2,Queuing caller for support agent)
exten = 201,n,Queue(support,nrt,,,120)
exten = 201,n,Verbose(2,Support agent did not answer call)
exten = 201,n,Voicemail(2...@ourvm,b)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
Use a dialplan to do what you want and dial that.
Originate a call to the first person and point it at context, exten, priority
that plays the sound file and then does a dial command to the number you want
them to talk to.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
You might be able to use local channels to do what you want.
As for the user asterisk runs as and the user the web server run as you can
maybe have both users belong to the same secondary group and gain the access
you need that way.
Partly depends on what exactly you are wanting to do.
--
Jim
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 10, 2010, at 7:35 AM, Eddie Mikell wrote:
Jim, and all:
Thanks for the response.
If I can repeat what you are saying: you don't have to define the multiple
lines in sip.conf?
Your SIP provider will limit
and the other to a voice tree that
allows the caller to specify the person's extension they want to talk with.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 29, 2010, at 1:33 AM, Ishfaq Malik wrote:
Philip
there; the second will not be skipped.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 28, 2010, at 5:49 AM, wassim darwich wrote:
Hi guys:
i need to set an extension in my dialplan in which it divert calls if the
extension contain specific series ,For example :
I
Do you mean you want
exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user ext-sip to that system's extension bob?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 28, 2010, at 10:50 AM, Aditya Kumar wrote:
Thanks Steve, I corrected
with allow from any 192.168.1.x address as an
example.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 27, 2010, at 4:31 AM, Vasiliy G Tolstov wrote:
Hello. I'm new with asterisk. Can you help me in this:
I have cisco sip phone (601) connected to asterisk server
I am not sure what you are asking here.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 27, 2010, at 6:16 AM, Vasiliy G Tolstov wrote:
В Втр, 27/04/2010 в 06:11 -0700, Jim Dickenson пишет:
In your sip.conf your permit line does not have an ip address to allow
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