2013/1/18 Danny Nicholas da...@debsinc.com
Since Gosub is technically an application, you should be able to modify
this snippet in features.conf
testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and
callee to play
;
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is connected (ie registered) to server A
-
Upgrading to the latest version didn't help. After about 30 minutes,
Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
Registered on Asterisk1.
It is something really amazing... Can you run sip show peers on each one of
the servers and post the response?
You said
The Dial events are created by app_dial. So long as you are using
app_dial to create your outbound channel, you should have that event.
Channel technology shouldn't matter.
I am using the same AMI method to start both calls.
Action: Originate
Channel: DAHDI/18/XX
or
Action: Originate
I was thinking of something similar, maybe using the URL field of the
queue() app as to point to an internal broker that will then link to the
message being used.
In theory one could do this for all kinds of traffic, including e-mails.
The part I don't really like is keeping an audio call open for
2013-01-23 18:20, Sebastian Arcus skrev:
I have an Asterisk server with one SIP trunk to a SIP provider. As my
server registers with the SIP provider, I don't have any SIP ports open
at my end to the Internet. However, I have the RTP ports open (as SIP
has some trouble with my NAT).
You could
Hi,
I am using Asterisk 11.2.0. Channel Event Logging (CEL) ist activated
and running. CEL entries are logged into an mysql database. So far so good.
I want to do some extra cel logging and try the following via an AGI-Script:
EXEC CELGenUserEvent test
In the asterisk logfile I can see the
I'm now getting these errors:
[Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-ba7 received frame with invalid timing info:
has_timing_info=1, len=0, ts=426891164, src=RTP
[Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-ba7
I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2)
did my AMI call
Action: Originate
Async: yes
Channel: SIP/testsystem/XXX
(calls from my machine over SIP trunk to another 11.0.2 box that has
a PRI card to make a call out to my cell)
and did not get a break.
Why is
2013/1/25 Dan Journo d...@keshercommunications.com
Upgrading to the latest version didn't help. After about 30 minutes,
Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
Registered on Asterisk1.
It is something really amazing... Can you run sip show peers on each
At a command prompt (not at the Asterisk CLI), if you run
dahdi_tool
and hit F1, what does it say?
This is what I see: http://i.imgur.com/je7qRHa.png
On Fri, Jan 25, 2013 at 8:20 AM, Richard Kenner ken...@gnat.com wrote:
I'm now getting these errors:
[Jan 25 09:19:01] WARNING[29877]:
I'm getting the following error, and none of us can figure out why:
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
Here is the code that generates it:
[scottsdale#queues-account]
Looks to me like ${prefix} contains nothing but two quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 11:09 AM
To: Asterisk Users Mailing List -
Error doesn't occur in 11.2.1
-- Executing [1260@default:1] Answer(SIP/sipuser-0001, ) in new
stack
-- Executing [1260@default:2] Goto(SIP/sipuser-0001,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing
Hi,
Let say that in a call center, callers are recognized and categorized in 4
priority levels (priority 1 for Very Very Important Personalities, 2 for
VIP, and so on) before entering a Queue.
How can you make sure a priority 2 caller is answered before priority 3
callers, for instance ?
I can
What version does the error occur on? I suspect more recent versions of
Asterisk removes extraneous quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, January 25, 2013 11:20
On Fri, Jan 25, 2013 at 9:20 AM, Eric Wieling ewiel...@nyigc.com wrote:
Looks to me like ${prefix} contains nothing but two quotes.
Which is as it should be unless they choose the Spanish option, but yeah,
maybe that's what is choking Asterisk.
We do this:
exten = _X.,n,Set(prefix=)
Don't do that. Set(prefix=) You are setting the prefix to have two quotes.
You WANT prefix to be empty.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013
On Fri, Jan 25, 2013 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote:
What version does the error occur on? I suspect more recent versions of
Asterisk removes extraneous quotes.
This is in 1.8.
Danny's test does support your theory. It looks like the var is being set
as the quotes,
2013/1/25 Alec Davis siva...@paradise.net.nz
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is
Am 25.01.2013 um 17:22 schrieb Olivier:
Hi,
Let say that in a call center, callers are recognized and categorized in 4
priority levels (priority 1 for Very Very Important Personalities, 2 for VIP,
and so on) before entering a Queue.
How can you make sure a priority 2 caller is answered
Where possible you should have a VM to try these things as needed. Where
not, it isn't too difficult to duplicate the contexts and do something like
this
[default]
.
.
Exten = 1260,1,answer
Exten = 1260,n,goto(test-context,s,1)
.
From: asterisk-users-boun...@lists.digium.com
2013/1/25 Michael Keuter li...@mksolutions.info
Am 25.01.2013 um 17:22 schrieb Olivier:
Hi,
Let say that in a call center, callers are recognized and categorized in
4 priority levels (priority 1 for Very Very Important Personalities, 2 for
VIP, and so on) before entering a Queue.
On Fri, Jan 25, 2013 at 9:31 AM, Danny Nicholas da...@debsinc.com wrote:
Where possible you should have a VM to try these things as needed. Where
not, it isn’t too difficult to duplicate the contexts and do something like
this
[default]
I do have a test VM, but I also have a
Am 25.01.2013 um 17:39 schrieb Olivier:
2013/1/25 Michael Keuter li...@mksolutions.info
Am 25.01.2013 um 17:22 schrieb Olivier:
Hi,
Let say that in a call center, callers are recognized and categorized in 4
priority levels (priority 1 for Very Very Important Personalities, 2
I've not tried to publish device state with XMPP yet but I've
discovered this issue
https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as
one asterisk server so hopefully, I won't be hit by the issue
above but have you met
On 01/25/2013 01:59 PM, Alec Davis wrote:
I've not tried to publish device state with XMPP yet but I've
discovered this issue
https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as
one asterisk server so hopefully, I won't be
1. Have 4 different queues, set penalty value and let each
caller enter one queue depending on its own priority.
Penalty isn't anything to do with the caller, it's to do with the agent.
We set round robin for our queues.
With penalty=0 for the main members of a queue, to service most of the
Not that this is an excuse or a valid workaround for
everyone, but I believe that issue won't apply if you're
using Asterisk 11 and res_xmpp.
res_jabber: yup, totally still a problem.
Hmm. We're using Asterisk 11, but I still think res_jabber.
Why havn't I changed to res_xmpp, I have no
I have just upgraded to asterisk 11 from 1.8
I have noticed that my Page command:
exten = 1,1,Page(SIP/101,diqA(local/intercom))
does not play the local/intercom sound to the conference.
according to the doc at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Page
, it
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