Asterisk 1.8.10.1~dfsg-1ubuntu1
A calls B.
B answers, intiaites an attendeded transfer to C.
C answers.
B hangs up.
A now hears ringing forever, until the call is terminated.
blind transfer does not have this problem.
What am I missing?
features.conf
[featuremap]
blindxfer = #1
atxfer = #2
If I remember correctly, INNODB offers row level locking while MyISAM
does not.
On 09/26/2012 05:18 AM, Thorsten Göllner wrote:
Am 26.09.2012 10:45, schrieb A J Stiles:
On Tuesday 25 September 2012, Matt Hamilton wrote:
Which one (InnoDB or MyISAM) is preferred for CDR as far as write
Asterisk 1.8.10.1~dfsg-1ubuntu1
Trying to build a simple announcement of the queue status. QUEUEHOLDTIME
is always zero. What am I doing wrong?
queues.conf
[general]
autofill=yes
shared_lastcall=yes
[StandardQueue](!)
musicclass=default
strategy=rrmemory
joinempty=no
leavewhenempty=yes
Callers:
1. SIP/mlcm800-0001 (wait: 0:52, prio: 0)
2. SIP/mlcx450-0003 (wait: 0:45, prio: 0)
Mitch
On 09/27/2012 03:16 AM, Lenz Emilitri wrote:
What do you get if you run a queue show sales?
l.
2012/9/26 Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net
) has taken no calls yet
Callers:
1. SIP/mlcx450-0003 (wait: 4:10, prio: 0)
On 09/27/2012 06:08 AM, Satish Barot wrote:
On Thu, Sep 27, 2012 at 2:39 AM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Asterisk 1.8.10.1~dfsg-1ubuntu1
Trying to build
I am also writing an AMI application that will allow management to see
the queue status from an external program and saw the same issues with
the AMI data. Using AMI I am able to get what I need from the
individual records for each queued call.
Mitch
On 09/26/2012 04:09 PM, Mitch Claborn
: Mitch Claborn mitch...@claborn.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 27 Sep 2012 09:20:08
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] QUEUEHOLDTIME
Warren - that coincides with what I am seeing. I guess it made sense to
someone, but it is not terribly useful to me.
mitch
On 09/27/2012 11:22 AM, Warren Selby wrote:
On Thu, Sep 27, 2012 at 9:15 AM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Satish I
I want to put a call me now button on the web site that will place the
request into an asterisk call queue and then when an agent picks up the
call in the queue, place the outbound call to the customer.
The following AMI command works, but it calls the customer first, before
an agent is
.
Mitch
On 09/28/2012 01:42 PM, James Sharp wrote:
On 9/28/2012 12:42 PM, Mitch Claborn wrote:
I want to put a call me now button on the web site that will place the
request into an asterisk call queue and then when an agent picks up the
call in the queue, place the outbound call to the customer
Looking for ideas and comments on my strategy for getting a bit of
custom data into the CDR. It seems to work OK, but I'm open to better
and/or more robust ways to do it.
Problem: get the customerid of the caller from our application into the CDR
Approach:
Before the Queue() command, save
Sam - can you send output from a top when your server is under load?
Just curious.
--
Mitch
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Asterisk 1.8 on Ubuntu
We store the configuration files in CVS. We have a development, QA and
production environments. 90% of the config files are the same across all
3 environments, but there are some differences in sip.conf and
extensions.conf (environment specific voip providers and/or
I'll give this a try today and post the results here.
Mitch
On 10/04/2012 02:30 PM, Ioan Indreias wrote:
Hello Mitch,
Hoping that the Queue application is not automatically Answering the
line (till an agent will do this) my suggestion is to switch between
who have to answer in order to
This is mostly working. See below. My only problem is being able to
set the caller ID on the outbound call to the customer. I've tried both
a queue connected macro and gosub (see below), and those both execute,
but the caller ID is not showing up correctly for the customer. I
assume this
Perfect! Thank you.
Mitch
On 10/05/2012 01:07 PM, Ioan Indreias wrote:
Hi Mitch,
Glad that it works for you.
Regarding the CallerID I suggest to set some the variables before the
actual Dial.
Something like:
Action: Originate
Channel: Local/s@callmenow/n
Context: to-customer
Exten: s
Asterisk 1.8
(a) We will have a group of 4 analog lines into a Digium card that will
be used for local calls. What is the best way to use those lines as a
pool for outbound calls? Can I use ChanIsAvail(), listing those 4
channels, and then use the first one returned?
(b) For emergency
Excellent. I'll give it a try.
(Now if I just didn't have to wait to get on-site where those lines are
to try it. Too bad there isn't a DAHDI emulator for SIP lines.)
Mitch
On 10/09/2012 10:48 AM, Richard Mudgett wrote:
There are lots of things documented in chan_dahdi.conf.sample. The
you expected to see?
Mitch
On 10/09/2012 12:40 PM, Shaun Ruffell wrote:
Minor correction below:
On Tue, Oct 09, 2012 at 12:32:44PM -0500, Shaun Ruffell wrote:
On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote:
(Now if I just didn't have to wait to get on-site where those lines
Here's what I came up with. Works find with the simulated DAHDI dynamic
local channels. I'll find out later in the week how it works with real
hardware.
[emergency-services]
exten =911,1,Goto(dialpsap,1)
exten =9911,1,Goto(dialpsap,1) ;
exten =999,1,Goto(dialpsap,1)
exten
Tomorrow evening I'll be at a customer site installing 2 Digum cards - a
4 port analog and 2 port T1. I'd appreciate any tips, resources and
links that you have that might help if we run into trouble. It will, of
course, be fairly late at night and relatively high pressure to get it
working,
I am a complete novice at T1's, etc. What else besides framing and
coding do I need to ask about?
Mitch
On 10/10/2012 10:41 AM, Jose P. Espinal wrote:
From my own experience, get sure that the Telco actually gives you the
*correct* information about the T1 (framing, coding, etc.).
There is actually only a single T1. When we ordered the card, customer
thought there were two, but found out later there is only 1.
Mitch
On 10/10/2012 11:50 AM, Steve Edwards wrote:
What is the relationship between the 2 Ts? NFAS? I've pissed away many
an hour trying to (remotely)
traveling for the next several hours, so apologies if I don't
respond right away.
Mitch
On 10/10/2012 10:34 AM, Mitch Claborn wrote:
Tomorrow evening I'll be at a customer site installing 2 Digum cards - a
4 port analog and 2 port T1. I'd appreciate any tips, resources and
links that you have
Last night we did a trial run. I am happy to report that both analog
and T1 lines worked well with the config files generated by
dahdi_genconf. Had a couple of minor issues that I'll ask about in
separate posts.
Of course when we got on-site, discovered that customer really has 6
analog
Setting up a group of analog lines to use for outbound emergency calls
(911). My current dial plan and debug output shown below. It appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single DAHDI
(analog)
Converting this customer from a MiTel system to asterisk. Discovered
that the inbound calls from the T1 are going to extension 366. (This
was mapped in the MiTel for some arcane purpose.) The dial plan I am
currently using is shown below. When loading the dial plan, I get this
warning:
The s extension did not catch the incoming call. It was only when I
added a specific 366 or the _. wildcard that I was able to capture the
incoming call.
Mitch
On 10/12/2012 10:18 AM, A J Stiles wrote:
If (and only if) all the extensions you are using in all your contexts are
numeric,
The built in (non-programmable) transfer button on the MiTel 5330 does a
blind transfer. Any ideas on how to make it do an attended transfer
instead? Instead of DTMF tones, it seems to send a SIP message to do a
transfer. I've been unable to find a way to change what it does.
--
Mitch
At the end of the output for core show channels verbose is a line that
reads 4 active calls. Does anyone know how that number is formatted
if there are more than 999 active calls? Will it have a comma or not?
--
Mitch
--
Dave Platt provided the following answer to a similar question of mine
last week. I was trying to use SoftHangup() to prempt a DAHDI line for
an emergency call. Here is his reply.
That may be due to a common characteristic of PSTN lines (at least,
it's common here in the U.S.)
By design,
Trying to set some CDR fields in the connected macro of a queue
command. None of the custom fields I set are stored in the database,
but I can set userfield and it does get set. I think that the macro
runs on the agent's channel, not the caller's, and this might contribute
to the problem.
Asterisk 1.8.10.1~dfsg-1ubuntu1
See dial plan code below. When I dial 123 from a phone in this context,
I simply get a busy signal. Why doesn't the i extension get
triggered? Console at verbosity of 10 only shows == Using SIP RTP
CoS mark 5.
[DockPhone]
exten =288,1,NoOp(Dock Phone)
I set logger.conf to
console =debug,notice,warning,error,verbose
and get the following output:
== Using SIP RTP CoS mark 5
[Oct 25 10:32:53] NOTICE[3501]: chan_sip.c:22622 handle_request_invite:
Call from 'Mitch295' (192.168.5.104:5060) to extension '123' rejected
because extension not
That does sound quite suspicious.
Mitch
It looks like you are seeing this issue that was fixed earlier
this month:
https://issues.asterisk.org/jira/browse/ASTERISK-20455
Richard
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A little more background will help. This is a phone that will be
outside on our receiving dock. When a driver lifts the handset, the
ObiTalk 110 dials 444 automatically. That all works fine and it rings
the phones that it should.
What I'm trying to do with the i extension is give a
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders with our back end systems. What I need to be
able to do is tie the orders taken to the specific CDR record that
reflects the call from which the order originated.
The typical/sample CDR table
Thanks Tony, this helps.
Mitch
On 10/25/2012 11:24 AM, Tony Mountifield wrote:
The 'i' extension is not used when entering a context. You can only enter
a context (with Dial(), Goto(), etc), at an extension that exists. If it
doesn't exist, the context cannot be entered.
The 'i' extension
DOCK_RECIPIENTS is a long list of 5+ SIP phones, so this won't work.
Mitch
On 10/25/2012 11:31 AM, Danny Nicholas wrote:
BOP! You don't need no stinkin I in this case! Just put this in front of
the Dial()
Exten = 444,2,Gotoif(${DOCK_RECIPIENTS} != 444]?i,1)
This catches anything they dial
...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Thursday, October 25, 2012 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to tie orders taken to specific CDR records
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders
Looking at the uniqueid, I get multiple records for some of them. Am I
getting more than one CDR record per call in some cases?
SELECT uniqueid, COUNT(*) FROM asterisk_cdr
GROUP BY uniqueid
HAVING COUNT(*) 2
Mitch
On 10/26/2012 08:34 AM, Bharat Lalcheta wrote:
Every CDR has
In our sales queue, we have wrapup time set to 15 seconds. When the
phones are really busy, the operators would like the ability to bypass
that 15 second wait and grab the next call in the queue. Is that
possible? How to accomplish?
--
Mitch
--
might offer a solution.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Monday, October 29, 2012 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Is there an Asterisk repository for Ubuntu that has recent versions
(e.g. 11)? The standard Ubuntu repository for Ubuntu 12.04 is stick at
1.8.
--
Mitch
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Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way
to determine the name of the calling context.
--
Mitch
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...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Tuesday, December 11, 2012 3:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking
asterisk 11.1
Documentation in cdr.conf for endbeforehexten reads:
Normally, CDR's are not closed out until after all extensions are
finished executing. By enabling this option, the CDR will be ended
before executing the h extension and hangup handlers so that CDR
values such as end and
We bypass this problem by storing business hours and holidays in a
database table. We use an ODBC call to return whether or not to play
the day or night greeting based on the database. We also store the
name of a custom greeting file to play.
The database is fairly easy to manipulate with
It would be nice (for me anyway) if the mailing list and forum were
combined. Google Groups does this nicely I believe.
Mitch
On 01/02/2013 08:53 AM, Eric Wieling wrote:
I don't use forums as my web browser can't automatically filter the messages
for me like my e-mail program can.
I
Asterisk 11
Occasionally we will have a partial power outage, or a piece of network
equipment will fail, and our queue agents who are on active calls with
callers will be disconnected from the caller. What I'd like to do is
capture those calls and put them back in the queue (at a high
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
Occasionally an agent will get a call (or more often a series of calls
in a row) where neither party can hear the other, or can only hear each
other sporadically. A MixMonitor
from my iPhone
On 18 mrt. 2013, at 19:31, Mitch Claborn mitch...@claborn.net wrote:
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
Occasionally an agent will get a call (or more often a series of calls in a
row) where neither
to fix the problem.
Mitch
On 03/18/2013 11:51 PM, Satish Barot wrote:
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Asterisk 11.1.0
Various soft-phone SIP clients
call center with 10-12 agents online at once using asterisk queue
, 2013 at 10:21 AM, Satish Barot
satish4aster...@gmail.com mailto:satish4aster...@gmail.com wrote:
On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn
mitch...@claborn.net mailto:mitch...@claborn.net wrote:
Asterisk 11.1.0
Various soft-phone SIP clients
call center
pc have more then one
network interfaces?
you can capture sip invites from soft phone by enabling debug on client
ip sip set debug ip ip of softphon upload sip trace then somebody can
halp you, should provide more information's.
On Tue, Mar 19, 2013 at 5:39 PM, Mitch Claborn mitch...@claborn.net
restrict it to use same range as in rtp.conf.
let me know if this solve you problem.
On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn
mitch...@claborn.net mailto:mitch...@claborn.net wrote:
We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3.
There is no NAT
on agent pc
Please provide your network setup for getting better idea of problem
On Mar 19, 2013 10:10 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
rtp debug on the calls that do not work correctly shows packets from
server to client only, none from client
should not use ports
below 1 because they are in use of other services like 5060 for sip.
On Tue, Mar 19, 2013 at 11:57 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
This was the client sending from port 39409 to server port 13429,
which is in the range. From
: 50b7a1e27bbb9f6043dfccff16d7be88@172.16.0.245:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.1.0
Content-Length: 0
--
Mitch
On 03/19/2013 07:18 PM, Mitch Claborn wrote:
Good point. I changed to 1 - 4.
Mitch
On 03/19/2013 06:17 PM, Asghar Mohammad wrote:
i had this problem
There is no firewall on the client.
I've compared the SIP messages between a successful call and a failed
call, and I can see no difference except for things like port numbers
and call IDs.
It only fails occasionally, not on every call.
Mitch
On 03/20/2013 01:16 PM, Asghar Mohammad wrote:
that works correctly. I can discern no
difference other than things like port numbers and call IDs.
Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe
that will make a difference.
Mitch
On 03/20/2013 02:09 PM, Matthew J. Roth wrote:
Mitch Claborn wrote:
Where is a good place
On 03/21/2013 09:48 AM, Matthew J. Roth wrote:
Mitch Claborn wrote:
Thank you for that most excellent post. I had guessed at most of the
SDP fields and meaning.
No problem. I actually like looking at SIP traces for some reason.
I have wireshark traces from the client and the RTP packets
is that the agent has to have one
headset for the phone and another for their computer (which they need
occasionally).
I get to go home on Saturday! The Digium phone deployment is simple
enough to manage remotely.
Mitch
On 03/22/2013 01:13 PM, Matthew J. Roth wrote:
Mitch Claborn wrote
I have seen that behavior also.
Mitch
On 03/28/2013 06:56 PM, Olivier wrote:
Hello,
I'm using Hanhup Handlers in a testing asterisk 11 system.
Within one such handler, I'm setting CDR values.
To me, it seems those changed CDR values are not saved in CDR back-end.
Can you confirm ?
Regards
I recently faced the same issue. I didn't find a way in Asterisk to do
what I wanted.
A good workaround is to use wireshark in batch mode (tshark) to trace
traffic to the IP address you are interested in. You should be able to
filter it to capture only SIP traffic.
Mitch
On 03/29/2013
/3/29 Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net
I have seen that behavior also.
Mitch
On 03/28/2013 06:56 PM, Olivier wrote:
Hello,
I'm using Hanhup Handlers in a testing asterisk 11 system.
Within one such handler, I'm setting CDR
Asterisk 11.1.0
One queue with strategy=leastrecent. (Full queues.conf below.)
Occasionally (several times today), a caller will get stuck in the
queue - there are operators available to take the call, but the caller
stays in the queue for a long time. Any idea what might cause this, or
with autopause and autopausedelay to see if that
will help.
Mitch
On 05/01/2013 01:11 PM, Mitch Claborn wrote:
Asterisk 11.1.0
One queue with strategy=leastrecent. (Full queues.conf below.)
Occasionally (several times today), a caller will get stuck in the
queue - there are operators available
Asterisk 11.1
We have a situation where one of our incomings POTS lines will not
answer. There are 2 lines configured by the Telco as a rollover group
(rings the line that is not busy) and they feed into a Digium AEX410 on
the server. The most recent time this happened, I did a
I am running 2.6.1. I'll give the 2.6.y a try.
Mitch
On 05/28/2013 10:53 AM, Shaun Ruffell wrote:
On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote:
Asterisk 11.1
We have a situation where one of our incomings POTS lines will not
answer. There are 2 lines configured
-20a479b/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o
Mitch
On 05/28/2013 12:37 PM, Mitch Claborn wrote:
I am running 2.6.1. I'll give the 2.6.y a try.
Mitch
On 05/28/2013 10:53 AM, Shaun Ruffell wrote:
On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote:
Asterisk 11.1
We have
The 2.6.y version installed without issue. A few test calls went OK.
Will leave it in and see how things go. The problem has been sporadic,
so won't know for a while if the issue is solved.
Mitch
On 05/28/2013 01:37 PM, Shaun Ruffell wrote:
On Tue, May 28, 2013 at 12:44:47PM -0500, Mitch
Asterisk 11.1.0
I'm trying to use the b subroutine of the Dial application so that I
can do some stuff with our internal applications that need to have
access to the called channel information. I can see that the subroutine
is being executed, but the arguments I pass don't see to make it to
On 08/02/2013 01:28 PM, Matthew Jordan wrote:
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Asterisk 11.1.0
I'm trying to use the b subroutine of the Dial application so that
I can do some stuff with our internal
We do something very similar.
Use the gosub parameter of the Queue application to call a subroutine in
the dial plan when the agent answers the call.
same =n,Queue(sales,tc,,sub-QueueConnected)
[sub-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
; whatever
Asterisk 11.1
Is it possible to catch the fact that an IP phone has died in the middle
of a call and do something with it in the dialplan?
Background: we run a small call center. Our agents sit in two groups,
with their IP phones running from 2 different switches. Every once in a
while the
I certainly agree that the first and best solution is to deal with the
hardware issues, and we've started working on that already.
I'll investigate the suggested Asterisk ideas and post here if anything
works for my purposes.
Mitch
On 11/08/2013 12:13 AM, Mikhail Lischuk wrote:
Mitch
The core show channels verbose command shows a calls processed
value. Mine is currently 1928273.
Exactly what does this figure represent? How is a call defined in
this context?
--
Mitch
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Asterisk 11.1.0 running on Ubuntu 12.04 64 bit
Dahdi
Digium T1 card
Occasionally, I will find an inbound call that just seems to be stuck,
usually in our after-hours menu portion of the dial plan.
This morning I had this one
core show channels concise
Short question: how to get control or notification (gosub, macro, AGI)
when a queue member's phone starts ringing due to an incoming call from
the queue.
Backround: Our phone operators serve both an asterisk call queue and a
queue for web chat support. I have a gosub on the queue that calls
I read somewhere that DPMA is not supported for Asterisk 12. Can anyone
confirm or deny that? If not supported yet, will it be? If so, when?
--
Mitch
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Is it possible (and advisable) to copy menuselect options from Asterisk
11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy?
--
Mitch
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Asterisk 12.4
I am seeing message Error opening file for reading: Permission denied
several times during the asterisk startup (asterisk -cv) but it
doesn't say which file. Is there a way to find out which file is having
trouble?
--
Mitch
--
));
return NULL;
}
Regards,
Paul
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Mitch Claborn
mitch...@claborn.net
Sent: Monday, August 18, 2014 1:14 PM
To: Asterisk Users Mailing List
I tried grep too.
No 3rd party modules - this is an out-of-the box download and build.
I'm guessing that some library function is being called to read a file
and the error is happening there?
Mitch
On 08/19/2014 02:33 PM, Matthew Jordan wrote:
On Tue, Aug 19, 2014 at 11:36 AM, Mitch
be because I'm starting asterisk as root. When I su to
asterisk first, then start it, those above disappear. Problem solved!
Thanks Steve!
Mitch
On 08/19/2014 03:39 PM, Steve Edwards wrote:
On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net
No, that's
Asterisk 12.5.0
DPMA 12.0_2.0.0
Ubuntu 12.04 64 bit
[2014-08-21 16:37:49] WARNING[5797]: phone_users.c:5236 set_and_process:
User SIP settings missing or invalid
I'm getting the error message above when DPMA is enabled, using a fresh
build but with my config files from Asterisk 11. Any idea
Asterisk 12.5.0
DPMA 12.0_2.0.0
Ubuntu 12.04 64 bit
WARNING[5797]: presencestate.c:147 ast_presence_state_helper: No
provider found for label CustomPresence
ERROR[5797]: pbx.c:4375 ast_func_write: Function PRESENCE_STATE not
registered
I only see these when DPMA is enabled. Any ideas what
loaded
Mitch
On 08/21/2014 06:55 PM, George Joseph wrote:
Make sure the func_presencestate.so module is being loaded. Did you
compile Asterisk yourself or are you using a pre-built from a distro?
On Thu, Aug 21, 2014 at 5:34 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net
Asterisk 12.5
I'm using AMI to initiate a call me now feature from the web site.
The AMI looks like:
Action: Originate
Channel: Local/s@callmenow
Context: dial-to-customer
Exten: s
Priority: 1
Async: true
Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/111222
Timeout: 99
Dial Plan:
Asterisk 12.5
The CoreShowChannel event (in response to the CoreShowChannels action)
no longer returns the Application field as it did in Asterisk 11. Is
this a bug or a feature?
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Mitch
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-- Bandwidth and Colocation
On 08/22/2014 02:47 PM, Matthew Jordan wrote:
Yup, that's a bug. When things got ported over to hit the cached
snapshots of the channels (as opposed to locking the live channel),
that field got missed. Please file a bug on issues.asterisk.org.
Thanks! Matt
Asterisk 12.5
I have a reproducible segfault using the MeetMe application. How do I
gather the necessary information (backtrace, core dump...) to submit a
bug report?
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Mitch
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Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous that is!
I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels
but I'm just not quite getting it.
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,Wait(1)
same =n,Playback(custom/callmenow-announce)
; do some more stuff
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,TKU(dial-to-cust-connect-sub))
Mitch
On 08/25/2014 11:43 AM, Joshua Colp wrote:
On 8/25/2014 1:33 PM, Patrick Laimbock wrote:
On 25-08-14 17:06, Mitch Claborn wrote:
Can
would work for the pager.
The script is very fast and does not interrupt the flow of the actual call.
Mitch
On 1/12/19 8:57 PM, Mitch Claborn wrote:
We have an overhead paging system that is working fine with our asterisk
16.1 server. I'd like to be able to push an announcement to the paging
I'm working on an asterisk upgrade to 16.1 and am remote from that
location. We use Digium phones there, configured with DPMA. From my VPN
I can connect to the server directly with the phone on my desk, but it
doesn't find the configuration server automatically since I'm on a
different
Thanks Ryan. Would you mind sharing snippets of your DAHDI channel
config and dialpaln?
Mitch
On 12/11/18 8:43 AM, Ryan, Travis wrote:
Yes it's very easy. Mine is using a simulated PRI over an ATT Flex line. I just
followed the many tutorials out there. I answer the call, then it takes 6-7
When building a new release, is it possible to copy the output of "make
menuselect" from a previous build directory? If so, what files need to
be copied? That would save some time in the upgrade process.
Mitch
On 12/11/18 4:11 PM, Asterisk Development Team wrote:
The Asterisk Development
I'm assuming that no one knows the answer to this.
Does anyone have fax detection successfully working? If so, can you
share your configuration?
Mitch
On 12/4/18 4:27 PM, Mitch Claborn wrote:
Asterisk 16 latest
DAHDI 3.0.0 latest
Excerpt from chan_dahdi.conf is shown below. I'm trying
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