Un-top-posting...
On Fri, 11 Mar 2011, satish patel wrote:
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script
doesn't working We have allpage.agi script for paging system on all
polycom 501 but after upgrade it broke. Any idea what is this error ?
[Mar 11 15:40:46] ERROR[31
Thanks for reply Steve,
I am not in office so i can't post script right now but will so once
reach home.
By the way that script working great in asterisk 1.2 my production
machine. But now I'm testing on 1.8.x and having issue which I
mentioned before.
This script is perl script and it
On Fri, 11 Mar 2011, satish patel wrote:
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script
doesn't working We have allpage.agi script for paging system on all
polycom 501 but after upgrade it broke. Any idea what is this error ?
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Friday, March 11, 2011 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Anyway to monitor SIP debug from originator and
t
Hi Everyone,
In order to make life easier and to do debugging easier I want to observe
"sip set debug originator" and "sip set debug terminator" on two different
putty screens. Trick is that originator calls the terminator. I can of
course put two separate calls and get sip debugs at different tim
Hey Guys,
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't
working We have allpage.agi script for paging system on all polycom 501 but
after upgrade it broke. Any idea what is this error ?
extension.conf
exten => 7770,1,agi(allpage.agi)
exten => 7770,2,meetme(7770
I am using 1.8.3 and changed enable=no on dnsmgr.conf - however I am
still getting log messages
for dnsmgr_lookup. I wasnt expecting that.
I have a server and a couple dedicated machines just running ALSA
connections.
I dont need any dns lookups for anything - who do I disable it?
Thanks
je
Thanks Jim! That's actually a great idea! I'm looking into that now!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Friday, March 11, 2011 12:12 PM
To: Asterisk Users Mailing List - Non-Com
Thanks for that Vladimir,
Despite being installed it doesn't seem to be recognized by asterisk:
*server55667*CLI> reload set show*
*No such module 'set'*
*No such module 'show'*
*server55667*CLI> reload set show ooh323*
*No such module 'set'*
*No such module 'show'*
*No such module 'ooh323'*
*s
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/
Thanks Danny! I took a look at the CDR data through AMI but it only throws an
event when the call is hung up. About the Unique ID... it looks like it stays
the same through the bridge. I've pasted the AMI output from a queue call
below. The only part that changes is the decimal increments.
Eve
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro
Sent: Friday, March 11, 2011 9:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do you handle queues with AMI?
Hey all,
I'm in the pro
Hey all,
I'm in the process of writing a few applications that are going to either
monitor the queue (number of calls, positions, etc) or respond to answering
a queue call (if you answer, a window pops up with info about caller, hold
time, etc.). I'm writing this in C# but language isn't import
Hello,
does anyone have a SIP trace for me where the SIPheader "Privacy: id" is
present ?? If so, what Asterisk version ?
Kind regards,
Jonas.
On 03/09/2011 06:43 PM, Bryant Zimmerman wrote:
Jonas
In my systems I have seen the Privacy: id when we do our testing but
it has been several mo
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
magnu...@inputinterior.se
Sent: Friday, March 11, 2011 3:59 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Automatically unpause a paused queue memeber -
badidea?
Try by reversing the line number of permit & deny
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, March 10, 2011 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
Hi ,
I am using asterisk SVN-branch-1.6.2 version
when i am making a call from SIP phone i found a warning of "Exceptionally long
voice queue length" .
When i search it on forum i found that This sounds like issue 15609 which has
been resolved newer versions of asterisk
https://issues.aster
I have some cases when I want to pause a queue member and automatically unpause
the queue member after a specified time.
Right now I am doing it by a AMI script, but was thinking if it is possible to
add a parameter to PauseQueueMember like,
PauseQueueMember([queuename],interface[,options[,reaso
Thanks, Smith.
Even keeping the file empty/ touch did not help.
Not sure still what we are missing.
--- On Tue, 3/8/11, Cassius Smith wrote:
From: Cassius Smith
Subject: Re: [asterisk-users] Cisco 7942G IP Phone firmware conversion from
SCCP to SIP.
To: "Asterisk Users Mailing List - Non-Com
Hmm disabled Woomera and everything seems stable.
Strange!
On Thu, Mar 10, 2011 at 11:46 AM, Peter den Hartog wrote:
> 1.8.0 :-), Nothing fancy just simple dialing/trunking.
>
>
> On Thu, Mar 10, 2011 at 11:31 AM, --[ UxBoD ]-- wrote:
>
>>
>> --
>>
>> My Asterisk 1.8
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