to additional documentation in addition to offering some
ideas as to why the above transaction might not work.
TIA
--
Jim Dickenson
mailto:[EMAIL PROTECTED]
CfMC
http://www.cfmc.com/
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Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jim Dickenson
Sent: Tuesday, October 14, 2008 6:58 PM
To: Asterisk User MailList
Subject: [asterisk-users] Help With AMI
I am trying to get
what I should look at to see why I am not getting a
busy instead of ringing?
Here are the definitions in sip.conf for the GXP280 and zoiper:
[dickenson]
type=friend
context=empl
nat=yes
host=dynamic
secret=password
callerid=Jim Dickenson 108
[EMAIL PROTECTED]
[GXP280]
type=friend
context=empl
secret=password
callerid=Jim Dickenson 108
[EMAIL PROTECTED]
[GXP280]
type=friend
context=empl
host=dynamic
secret=password
callerid=GXP280 109
[EMAIL PROTECTED]
busylevel=1
I am wondering if anyone has been able to get a busy when calling a SIP
phone. I seem to just get a no answer
contexts so it was a bit confusing as to which context the extension was in
at some given point in time.
After changing things to match contexts stuff worked as expected.
--
Jim Dickenson
mailto:[EMAIL PROTECTED]
CfMC
http://www.cfmc.com/
From: Marco Signorini [EMAIL PROTECTED]
Reply
=dynamic
secret=mypwd
callerid=GXP280 109
[EMAIL PROTECTED]
busylevel=2
[dickenson]
type=friend
context=empl
nat=no
canreinvite=no
host=dynamic
secret=mypwd
callerid=Jim Dickenson 108
[EMAIL PROTECTED]
Looking at the SIP debug transaction I do not see my public IP address at
all. There must be some
will also use STUN. Configure the
³Use Random Port² to ³NO² when completing Direct IP calls.
--
Jim Dickenson
mailto:[EMAIL PROTECTED]
CfMC
http://www.cfmc.com/
From: Valentin Bud [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
to continue talking
while listening to the sound file?
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Jim Dickenson
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might want to perform with the original two legs.
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Jim Dickenson
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From: Danny Nicholas [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thu, 20 Nov 2008 16:46:54 -0600
I have played with both call-limit=2 and busylevel=1 and have yet been able
to get a busy returned from a call attempt. If someone actually has this
working some insight into how you are setup would be much appreciated. I can
cause a channel unavailable but not a busy.
--
Jim Dickenson
mailto
not specified
Is anyone else seeing anything like this?
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})
exten = _1XX,n,Hangup()
exten = _2XX,1,NoOp(Calling to server 2)
exten = _2XX,n,Dial(IAX2/f3t2/${EXTEN})
exten = _2XX,n,Hangup()
--
Jim Dickenson
mailto:[EMAIL PROTECTED]
CfMC
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From: Tóth Csaba [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing
There was a problem introduced in 1.6.0.2, as I recall, that is fixed in the
current release candidate for 1.6.0.3.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Christian christia...@runbox.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
If the diversion takes place in asterisk then the dialplan can set a
variable before it diverts and then use that variable at the destination of
the diversion.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Julian Lyndon-Smith aster...@dotr.com
Reply-To: aster
I had the same problem, I think with that version. You need to use a more
current rc version. I think I am using rc2 but rc3 has been released as I
recall.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Andrew Nowrot andrew.now...@gmail.com
Reply-To: Asterisk
Variable: WhoHear=Other-channel
Variable: WhatToPlay=dont-know-who-sent
ActionID: callE1324
Async: true
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What I have done in the past to set the password for root is to boot in
rescue mode and edit /etc/shadow setting the password to some know value
from another system.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: David @ULC ucoms2...@gmail.com
Reply-To: Asterisk
If you are running a rpm based system you should be able to do ³rpm Va² and
see if anything has change from when all the stuff was installed.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Matt Watson m...@mattgwatson.ca
Reply-To: Asterisk Users Mailing List
as if I use this extension twice at
the same time there would be a conflict. What is the correct way to handle
this.
This is while running Asterisk 1.4.23
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Jim Dickenson
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.
The only other conditions that set running to -1 is if you hang up the
channel or res = (f-frametype == AST_FRAME_DTMF) ? f-subclass : 0; sets
res 0. As I do not know what I can type on the keypad to cause that I do
not know how to exit the chanspy application.
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Jim Dickenson
mailto:dicken
not particularly want to do.
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Jim Dickenson
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Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Stopping chanspy followup
Jim Dickenson wrote:
I am still trying to figure out a reasonable way to exit the chanspy
application in a dialplan.
For the most part I understand how things are working and there is one
= QueueAnswer,12(ForceHangUp),Hangup()
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at least one problem.
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not found. The
extension displayed has uppercase characters so the case is not lost.
If I change the definition and use to somename then things work as expected.
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in 1.6, would resolve I problem I submitted a
patch for a few days ago to 1.4.23.1.
It might resolve the problem but I can tell as this loop problem occurred
during my testing.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
*/
+body-used = 0;
for (x = 0; x m-hdrcount; x++) {
if (strncasecmp(UserEvent:, m-headers[x], strlen(UserEvent:)))
{
ast_str_append(body, 0, %s\r\n, m-headers[x]);
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
and CHANNEL
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Jim Dickenson
mailto:dicken...@cfmc.com
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From: Mindaugas Kezys mke...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sun, 22 Feb 2009 11:42:41 +0200
To: asterisk-users@lists.digium.com
what I might look at to find what is causing this
problem?
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the past that are different now.
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From: Brent Davidson br...@texascountrytitle.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thu, 26 Feb 2009 15:18:14 -0600
You specify a channel. Just specify the channel on the other leg of the
call.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Jerry Geis ge...@pagestation.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date
if there is a setting I can set to increase the 30 second timeout
that is occurring to some higher value.
Does anyone know why the PRI line does not want to ring for more than 30
seconds?
TIA
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Kevin P. Fleming kpflem...@digium.com
Organization: Digium, Inc.
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 02 Mar 2009 14:41:51 -0600
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Michael mich...@networkstuff.co.nz
Reply-To: mich...@networkstuff.co.nz, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Date: Fri, 6 Mar 2009 15:52:20 +1300
To: Asterisk
In version 1.6.0.x if you enable dialplan events in AMI you then get a
packet for each dialplan step executed. You should be able to capture that
data and generate records in a database.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: nik600 nik...@gmail.com
Reply
What version of * are you running?
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Jim Dickenson
mailto:dicken...@cfmc.com
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From: Sebastian s...@adinet.com.uy
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thu, 12 Mar 2009 18:41:33 -0300
To: 'Asterisk
...@ourvm,u)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
exten = 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten = 201,n,Queue(support,nrt,,,120)
exten = 201,n,Voicemail(2...@ourvm,b)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
If you were using 1.6 then you could do it in one queue with the new queue
rules, at least as I read the docs.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Christopher Aloi chris.a...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
I am guessing that your dialplan has a dial command to actually do the
dialing. What we do is pass a variable, like dial-timeout, and then use
${dial-timeout} as the time out in the dial command.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Singh Saimbhi
?
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} and ${MEMBERNAME} have info about the agent that answered
the call.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Luis Morales faston...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tue, 14 Apr 2009
=2
echocancel=no
signalling=fxo_ls
channel = 145-164
context=from-cbfxo
group=3
echocancel=no
signalling=fxs_ls
channel = 165-168
Held figuring this out would be appreciated!
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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TDMV_HW_DTMF= YES
-[w6g1]
+[w7g1]
ACTIVE_CH= ALL
TDMV_ECHO_OFF= NO
TDMV_HWEC= YES
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Lyle Giese l...@lcrcomputer.net
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
When I compile the source code on a 64bit system it made a 64bit executable.
I did not have to do anything special. This was on a CentOS 5.3 system.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Klaverstyn, David C david.klavers...@intergraph.com
Reply
http://www.ex-parrot.com/~pdw/iftop/
You can use http://www.rpmfind.net/ to find where stuff you are looking for
can be found.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Dave Walker d...@damcconsulting.com
Reply-To: Asterisk Users Mailing List - Non
tone.
Any ideas as to what might cause this behavior?
Thanks for any ideas!
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Jim Dickenson
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-users] Sangoma a104d and channel banks
2009/5/7 Jim Dickenson dicken...@cfmc.com
I have * 1.6.0.9 with dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.2, libpri
1.4.10 and wanpipe-3.4.1 running on CentOS 5.3 64bit.
I have 2 ports of the a104d configured for use with PRI lines and 2 ports
configured
-users] Sangoma a104d and channel banks
2009/5/7 Jim Dickenson dicken...@cfmc.com
From: Grygoriy Dobrovolskyy megaho...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thu, 7 May 2009 12:20:07 +0200
To: Asterisk Users Mailing
I have more info now.
I have used AMI to watch events, including DTMF events. The phone comes off
hook and I see a line on the * CLI saying the line is off hook. I then touch
a key and I see the DTMF event in the AMI. At the same time I see a message
on CLI that says hangup the line.
--
Jim
- TIMEOUT FULL JOINEMPTY LEAVEEMPTY JOINUNAVAIL LEAVEUNAVAIL
CONTINUE
exten = queue_answer,n,UserEvent(AfterQueue,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_AgentToUse} ${CfMC_DialInfo}
${CfMC_QueueToUse} ${QUEUESTATUS})
exten = queue_answer,n(ForceHangUp),Hangup()
--
Jim Dickenson
this, as well as *79, and turns DND on and off for the channel.
Is there some way to not have chan_dahdi intercept any DTMF digits?
I still have a problem in that when I enter any DTMF digit, like 1, the line
immediately hangs up.
TIA
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
I have DTMF detection enabled. I will check if overlap makes a difference.
Right now I have it set to no.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Steve Davies davies...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
for SIP I did not associate my problem with unknown extensions.
To find this the console line in logger.conf got debug added to it and I
cranked up the debug level and eventually saw an error about no matching
extension.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From
} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
What this does is dial an extension to play what I want to play and then
bridge in a ChanSpy with whisper so the extension I want to hear the sound
can listen in.
This seems to do the trick for me.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
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://www.cfmc.com/
From: David Anthony O Reilly oreil...@tcd.ie
Reply-To: Asterisk Users Mailing List - Non-Commercial
to
press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: David Anthony O Reilly oreil...@tcd.ie
Date: Sat, 16 May 2009 14:54:00 +0100
To: dicken...@cfmc.com
Cc: asterisk-users
port?
Would I then have 24 voice channels that I could manage for the 24 phone
sets?
The only thing I know about the T1 is that it uses wink start signaling.
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Jim Dickenson
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I connected without Proxy filled out.
I used name and not IP address in Realm and Server
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Jim Dickenson
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From: César Sequeira cesar.m.seque...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
I think it is a DID trunk. I am having problems getting the clients telco to
tell me much about the T1. For sure 24 analog channels in a single T1.
Would I be able to use this type of T1 with a Sangoma A102de?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Wilton
notes about the problem of rare mid conversation hang ups so I
will watch for that.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Jeff LaCoursiere j...@jeff.net
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date
} ${CfMC_DialInfo} ${DIALSTATUS})
exten = dial_number,n,Hangup()
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: John Regal jre...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wed, 3 Jun 2009 14:38:09
Xorcom makes Astribank devices that have two USB connections so one can go
to one system and one can go to a backup system.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Frank Bulk frnk...@iname.com
Organization: iName.com
Reply-To: frnk...@iname.com, Asterisk
???
There are no messages in the full log file before these line since 21:43 on
6/16/2009.
I am running asterisk 1.6.0.9, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2,
libpri-1.4.10 and wanpipe-3.5.2.
The PRI line is plugged in to a Sangoma A102de.
Any hints would be appreciated.
--
Jim Dickenson
=no
;Sangoma A102 port 1 [slot:4 bus:7 span:1] wanpipe1
group=1
context=
switchtype=national
echocancel=no
signalling=pri_cpe
channel =1-23
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Darryl Dunkin ddun...@netos.net
Reply-To: Asterisk Users Mailing List
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim
Dickenson
Sent: Wednesday, June 17, 2009 16:04
To: Asterisk User MailList
Subject: Re: [asterisk-users] What causes this error?
/etc/dahdi/system.conf has this:
loadzone=us
defaultzone=us
#Sangoma A102 port 1 [slot:4 bus:7 span:1
for
/usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
Anyone else seeing this?
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I download the tar.gz file and expand it, without error. I am not sure how I
could not have a complete download.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Tzafrir Cohen tzafrir.co...@xorcom.com
Organization: Xorcom*
Reply-To: Asterisk Users Mailing List
(cfmc_cdi_private, h, 2)
exited non-zero on 'Local/retry_num...@cfmc_cdi_private-4ff4;2'
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I am using version 1.6.0.x and you can do ³core show application dial² at
CLI to see info about the dial command.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: John Regal jre...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
http://bugs.digium.com/view.php?id=14595 has a patch to add a new class,
bridge, so you get less events in AMI. This is for 1.6.0.x. It will give you
an idea of what needs to be changed in order to make the call class of
messages more granular.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
It does which is why it was not included in a release code set. The patch
could be changed to do an OR type compare for the bridge class. I have
changed my implementation to use only user events for everything that I now
need so I did not pursue this patch.
--
Jim Dickenson
mailto:dicken
names are upper case so you need to use upper case when trying to
list them!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Steve Totaro stot...@asteriskhelpdesk.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date
on would help.
TIA
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
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of
another one with it. I guess I thought that extensions were discrete units.
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Jim Dickenson
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CfMC
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From: Stephen Davies stephen.l.dav...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
= 212555,1,Goto(companyb,menu,1)
; then separate contexts for each company:
[companya]
extern = menu,1,.
extern = menu,n,.
[companyb]
extern = menu,1,.
extern = menu,n,.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
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From: Myles Wakeham my...@techsol.org
=myusername
defaultuser=myusername
secret=mypass
context=sip-out
disallow=all
allow=ulaw
allow=alaw
Then in extensions.conf I have this:
exten = _9XX.,1,SIPAddHeader(No-Answer: true)
exten = _9XX.,n,Dial(SIP/gizmo5/${EXTEN:1},20)
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
I was able to get a VMWare Fusion CentOS 5.3 with Asterisk 1.6.0.9
talking to a Xorcom Astribank on my MacBook. I could connect a POTS
line to an FXO port and a phone to an FXS port and make calls.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 7, 2009, at 9
will exten = _13[0-4]X,1,Goto(12${EXTEN:3},1) work for you?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 17, 2009, at 9:58 AM, Gabriel Ortiz Lour wrote:
Hi all,
When I have 2 masks that would like to execute the same logic,
there is the way to use
1250628579|MANAGER|S1140|Agent/265|ADDMEMBER|
Above is an example of a line from the file. Is there an option to
have this date/time output in a way that can be read without
conversion? The date/time in full and messages is formated so it can
be read directly.
--
Jim Dickenson
$
exten = _Queue.,n,GotoIf($[${DEVICE_STATE($
{AgentDevice})}=NOT_INUSE]?DIAL$
exten = _Queue.,n,Busy()
exten = _Queue.,n,Hangup()
exten = _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten = _Queue.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 21, 2009
In voicemail.conf you can set serveremail=asterisk so some other
address and this will be used as the sender's email address, at least
as I understand things.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 24, 2009, at 9:50 AM, Michelle Dupuis wrote:
Do
there.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 9, 2009, at 7:29 AM, Patrick wrote:
Thanks a lot Danny but isn't it another production proof way of
doing this ?
I've given a very simple example but in production, I can call up to
10 destinations
ChanSpy can do that.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 9, 2009, at 9:31 AM, Valter Nogueira wrote:
Hi,
I would like to listen an ongoing conversation and whisper to
operator.
Is that possible in asterisk?
thanks,
Valter
You have your wireless access point password protected I am hoping.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 12, 2009, at 4:04 PM, randulo wrote:
François,
Just to be clear, I am not on Free at all but with our mutual friends
at Acropolis Télécom. I
Priority: 1
Variable: CfMC_ActionID=GetInfo
ActionID: GetInfo
Async: true
This dials a local channel and connects the call with another local
channel.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 2, 2009, at 8:15 AM, Danny Nicholas wrote:
The 1 (1,1; 1,2; 1,3) needs
You need to yum install kernel-devel'
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 6, 2009, at 8:40 AM, David Home wrote:
Hello,
Help need to solve Problem...kindly let me know how to solve the below
problem now i have CentOS 5.3 (Final)
I am new
Easier than poking around the source code is to add a DumpChan() step
to any area of your dialplan to see all defined variables.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 30, 2009, at 8:51 AM, Danny Nicholas wrote:
This thread taught me my Asterisk Lesson
})}=NOT_INUSE]?
DIALIT)
exten = _Queue.,n,Busy()
exten = _Queue.,n,Hangup()
exten = _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten = _Queue.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 2, 2009, at 8:44 AM, Lenz Emilitri wrote:
We were thinking about doing
Variable: CfMC_DoAMD=No
ActionID: CallAndQueue
Async: true
and things work as expected. This is in recent versions of 1.4 and 1.6.0.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 3, 2009, at 4:22 PM, Matthew Edmondson wrote:
Hi guys I seem to be having
to do any summary reports directly in CLI.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 4, 2009, at 11:51 AM, Mike wrote:
Or am I misunderstanding the displayed information? What I was looking for
was a list of channels that shows which one is currently busy
You do not need any special sendmail options/settings/configuration to use it
with Asterisk. If you are talking about sending voicemail notices then you just
need to point mailcmd in voicemail.conf to a command that can send the message.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
sound file to play.
SendDTMF or Playtones should be able to inject tones.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 12, 2010, at 4:19 PM, Steve Murphy wrote:
Dave--
I remember adding a feature a long time ago for snoms, to the source code,
to send dtmf
asterisk to it in its current state will likely be a good solution.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 20, 2010, at 4:50 PM, Gergo Csibra wrote:
Thursday, January 21, 2010, 12:53:09 AM, Jeff wrote:
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday
What I do is have this:
exten = h,1,UserEvent(DialHungUp,ActionID:${CfMC_ActionID} ${UNIQUEID}
${CHANNEL} ${CfMC_AgentToUse} ${CfMC_DialInfo} ${CfMC_QueueToUse}
${HANGUPCAUSE} ${DIALSTATUS})
And then in AMI I know hangupcause and dialstatus.
--
Jim Dickenson
mailto:dicken
I do not remember and issues we have between 1.4 and 1.6. When going to your
pastebin I get this:
Sorry, an error has occurred. Reason: That is an invalid ID, or the post has
expired.
Can you post what your ami packets contain?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http
In logger.conf do you have verbose and debug on the console line? If not add
them and do logger reload.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 3, 2010, at 2:13 AM, Tim Culhane wrote:
Hi,
For some reason I can't get Asterisk to produce debug
] -- Executing [...@context:2]
ExecIf(IAX2/trunk-12353, 0?Set(DB(conf//haveadmin)=no)) in new stack
Any ideas as to what might cause this behavior?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com
At an Asterisk CLI use the command manager show commands.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 5, 2010, at 1:50 AM, Peter Childs wrote:
Is there a list of input's / out puts from the management API together
with there parameters, there meanings
trunk to the system with the PRI
line.
Is there some setting in some conf file that allows or disallows this behavior?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
--
_
-- Bandwidth and Colocation
:testing
Var-02:eventmemberstatus
Value-02:yes
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 5, 2010, at 8:10 AM, Anahi Ludueña wrote:
Hi, I'm executing some commands using AMI... I suppose the log is saved in
some place, but I don't know where... where
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