we should be able to
take advantage of the live migration tools, and avoid downtime.
Definitely seems worthwhile.
Robert Jackson
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Maik Hassel wrote:
-A INPUT -s 192.168.1.0/255.255.255.0 -p udp -m udp --dport 5060 -j ACCEPT
You also have to allow the rtp streams through. You can configure the
range of ports for this in rtp.conf, but the defaults are UDP ports
1 - 2.
Hope this helps,
Robert Jackson
Brian Litzinger wrote:
Just no DTMF with calls via livevoip.
I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
Try updating to the latest stable version (1.0.7). We are using a
number of LiveVoIP inbound toll-free's and our DTMF is working well.
Robert Jackson
Matthew Boehm wrote:
Since t38 is seperate from SIP, you basically need a chan_t38 right?
-Matthew
That is my understanding.
Robert Jackson
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the manager interface.
There is more information on the wiki:
http://www.voip-info.org/wiki-Asterisk+monitoring
Good luck,
Robert Jackson
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Samuel Tardieu wrote:
Hi.
In zapata.conf, if I have:
foo=bar
context=line1
channel = 1
context=line2
channel = 2
Does foo=bar apply to channel 2 as well?
Yes.
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luck and welcome,
Robert Jackson
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} and
pass them along with the 02 that is needed.
I did not test this but I believe that is how it works. Check out the
what the wiki says about substrings here:
http://www.voip-info.org/wiki-Asterisk+Variables
Good luck,
Robert Jackson
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the KISS method ;)
FYI - (I just suggested that he use ${EXTEN:-8})
Robert Jackson
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to before the changes were committed. Also,
you could watch the -cvs list to see when the fixes are made
for res_config_mysql.
Good luck,
Robert Jackson
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We use LiveVoIP. They are very good to work with, and have
great pricing. Their website is livevoip.com, but I would
contact them via e-mail first.
Good luck,
Robert Jackson
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http
post. A lot of people belong to both lists
and as a result now have to look at the same e-mail twice.
I responded to -biz originally.
Robert Jackson
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opinions on that.
* Analyzing what specific functions/features you need.
Only then can this list help you.
Also, please try to research your questions via the wiki and
google. Both are an invaluable tool, and many topics have
already been discussed.
Good luck,
Robert Jackson
)
There:
https://sourceforge.net/tracker/index.php?
func=detailaid=746083group_id=29880atid=541465
Added IAX ping :)
roy
Thank you very much. That is exactly what I was looking for
as well.
It works great for us.
Thanks again,
Robert Jackson
or end up doing.
Robert Jackson
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documentation:
http://www.voip-info.org/wiki-Asterisk+QoS
http://www.voip-info.org/wiki-Asterisk+config+iax.conf
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Make close note of the TOS flag in both iax.conf and sip.conf.
Good luck,
Robert Jackson
,
Robert Jackson
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.
Robert Jackson
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not to use HTML e-mail on the list. ;)
Hope this helps,
Robert Jackson
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,
Robert Jackson
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!
Robert Jackson
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= 24XX,1,SetGroup(${EXTEN})
exten = 24XX,2,CheckGroup(1)
exten = 24XX,3,Dial(SIP/${EXTEN})
exten = 24XX,103,Busy
This will also keep the ACD calls from going to a persons voicemail box,
which would probably happen if your queue member didn't answer.
Hope this helps,
Robert Jackson
will have to test it to see what happens
when the caller hits a button not defined.
If you do test it please let us know how it behaves when an extension
is entered, but not specified in the context.
Hope this helps,
Robert Jackson
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for reaching an agent at an extension and another set for
simply reaching the extension outside of an ACD context.
This is how we have it setup and it seems to work pretty well.
Hope this helps,
Robert Jackson
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, and good luck.
Robert Jackson
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,
Robert Jackson
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= _1XXX,1,Ringing
exten = _1XXX,2,Dial(SIP/${EXTEN},15,t)
Or something like it. The only problem that I can see with that
is if your agent doesn't answer. The caller will hear ringing,
but it will just go back to music on hold.
Just an idea,
Robert Jackson
that you specify.
This has been my experience. I am not sure if it was designed this
way on purpose, but it seems to work this way for me nonetheless.
Good luck,
Robert Jackson
-Original Message-
From: Nathan Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 27, 2004 9:40 PM
this can be a
problem. The solution to all of this was to
SetVar(TRANSFER_CONTEXT=context that can transfer) before
you call Dial(device,20,t). Then * uses the context that
you specified to use for transfers.
Fixed a similar problem for us,
Robert Jackson
like to
see if anyone else has gotten it figured out.
Robert Jackson
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)
Check out the current config/extensions.conf.sample. This is exactly
How the relatively new dialstatus variable is used.
Robert Jackson
(Excerpt from extensions.conf.sample):
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here
-Original Message-
From: Robert Jackson
Sent: Friday, October 22, 2004 12:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi in stable? (New subject)
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED
to answer it.
(Also, posted to the bug tracker.)
Hope this helps,
Robert Jackson
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referenced it.
On a CVS-HEAD-10/20/2004 machine following the same procedure I do not
receive the error.
Any ideas?
Thanks,
Robert Jackson
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including bugfixes and anything else that is added to the 1.0
branch. Using cvs without the -r v1-0 gets you head.
Good luck,
Robert Jackson
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DUNDi made /. Check it out at:
http://www.dundi.com
Yet, another great idea!! Thanks Mark!!
I wish it was in v1.0, but I guess I'll have to update to head.
Robert Jackson
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) and added them to the linked
list which made the mwi work.
Just my $.02 worth,
Robert Jackson
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and
whatnot so that the main info could be added to the db when a
lookup is done and the MWI's could be generated from that, but
I didn't have the time to really look into it fully.
I am sure that there are better ways around this, but these are
just the ones that I came up with.
Robert Jackson
complex
especially if you have many remote extensions.
Just a couple of ideas,
Robert Jackson
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us to accomplish
the same goal. This even had an unexpected side effect for us over
Our previous system which preformed like you wanted: it kept our
Receptionists from dealing with the same call nearly doubling
their effectiveness.
I hope this helps,
Robert Jackson
within the dialplan.
Hope this helps,
Robert Jackson
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I forgot to add a link to the system command:
http://www.voip-info.org/wiki-Asterisk+cmd+System
-Original Message-
From: Robert Jackson
Sent: Sunday, September 26, 2004 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialplan
indefinately every 5 sec we check to see if there is
anyone logged in. Then we create a variable
(AGENTSLOGGEDIN) which is either 0 or 1. Then we check
the status of that variable from the dialplan to see
if we should place calls in the queue.
Seems to work pretty well for us.
Robert Jackson
me know if we are on the same page.
Robert Jackson
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was able to fix the
problem by executing
Answer before I entered any other applications.
Using your previous example:
exten = 2688,1,Answer
exten = 2688,2,Wait,3
exten = 2688,3,MeetMe,|Mps
exten = 2688,4,Hangup
Hope this helps,
Robert Jackson
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call.
Is there a way to automatically do the post call processing
like the 'm' option like when specifying the use of the
monitor command?
Try monitor-join=yes in queues.conf after the monitor-format line.
This seems to join the two files together for me.
Hope this helps,
Robert Jackson
-info.org/wiki-Asterisk+config+agents.conf
* http://www.voip-info.org/wiki-Asterisk+agents
* http://www.voip-info.org/wiki-Asterisk+call+queues
*
http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+co
mmands (Under Queue and ACD management)
Hope this helps,
Robert Jackson
/wiki-Asterisk+cmd+AgentCallbackLogin
Good luck,
Robert Jackson
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populating from a database do you have
[EMAIL PROTECTED] in your voicemail.conf?
Hope this helps,
Robert Jackson
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and whatnot
will be found, but I have found that CVS Head is pretty stable.
Untill RC1 was released I was using the latest CVS for our main
production system.
Thanks for your help!
Glad to be able to help.
-Ryan
Robert Jackson
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/4/2004.
Thanks for your help,
Robert Jackson
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you can have them hit one to leave a message or two
to receive a callback, etc...
I appreciate all the help.
No Problem, I hope this qualifies.
Warm Regards
Shad Mortazavi
Robert Jackson
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that it is for use with IAX, but I could be wrong.
Robert Jackson
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be in configs/features.conf.sample unless you
have run make samples in which case this file is copied to
/etc/asterisk.
Robert Jackson
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this helps,
Robert Jackson
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to voicemail.
Thanks,
Robert Jackson
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on skill level and or circumstances.) I
personally have three asterisk boxes running on Gentoo 2004.1 with great
success.
Hope this helps,
Robert Jackson
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If I am not mistaken that is similar to what VoicePulse Connect just
changed to. In there e-mail on how to configure it they have nearly the
same senario.
-Original Message-
From: Chris Shaw [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 28, 2004 6:14 PM
To: [EMAIL PROTECTED]
working, but we need to address the voicemail issue. I will
open a bug if this is not just something on my end.
Anybody else having issues?
Robert Jackson
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I am successfully doing this using the ast_data patch. Everything seems to be working
very well. You can download it at http://svn.asteriskdocs.org/res_data.
-Original Message-
From: Carlos Chavez [mailto:[EMAIL PROTECTED]
Sent: Monday, July 26, 2004 6:11 PM
To: Asterisk
Subject:
with this please let me know.
Thanks,
Robert Jackson
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have it working like this:
exten = 700,1,Answer
exten = 700,2,Echotest
exten = 700,3,Hangup
Hope this helped,
Robert Jackson
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how I was able to specify which context to use
when the user presses #. I haven't been able to find it on the wiki or
via google. Does anyone know off the top of their head?
Thanks,
Robert Jackson
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on my 7905's or on my 7960's. My assumption
would be that I am still missing something, but at this point I can't
figure it out. I have recently seen a message that Notify is not
working properly with CVS HEAD.
Thanks for you help in advance.
Robert Jackson
That would certainly make sense, but I am not sure how to set an Agent's
priority. The only information that I have been able to find is setting
a QUEUE_PRIO value when queuing the calls (New as of July 2004).
Thanks,
Robert Jackson
-Original Message-
From: Steve Hanselman [mailto
to leastrecent it is basically
only ringing the first two or three in order of agent number. This is
very bizzare. I was thinking that it was just something in my config,
but I just can't find out what it is.
Thanks for the help,
Robert Jackson
Pro-Medical, Inc.
-Original Message-
From: Chris
;context=qout
timeout=15
retry=5
maxlen=0
member=Agent/@2
[patient_q]
music=default
announce-holdtime=once
announce-frequency=90
strategy=leastrecent
;context=qout
timeout=15
retry=5
maxlen=0
member=Agent/@1
Thanks for your help,
Robert Jackson
Pro-Medical, Inc
That worked great! Thanks for the help. Any ideas on the uneven
distribution problems? Right now the agent with the lowest agent number
is getting 45% of the calls. She is going crazy!
Just trying to figure out what I screwed up.
Thanks,
Robert Jackson
Pro-Medical, Inc.
-Original
= _. So I guess my question
is what am I doing wrong? I know that * has to be able to interpret
this information. I am assuming that something is wrong with my
configs.
Thanks for the assistance,
Robert Jackson
zapata.conf
---
[channels]
usecallerid=yes
rxwink=300
echocancel=yes
-Original Message-
From: chouck [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 01, 2004 6:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Config Files
Im having a trouble understanding the config files setup even with some
documentation ive read such as the handbook, maybe im
I am pretty sure that it does not require any timing devices for use
with the VoiceMail2 app. I believe that I setup my first * box as a
simple test between two SIP phones with voicemail and it worked
properly.
Good luck!!!
Robert Jackson
-Original Message-
From: Paul Mahler [mailto
);
$AGI-exec('Queue', 'unverified-patientq');
exit(0);
}
$sth-finish;
$dbh-disconnect;
Exit;
Any help would be greatly appreciated.
Thanks,
Robert Jackson
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at different areas of the file to apply to all
or just some of the agents.
Hope
this helps,
Robert
Jackson
-Original Message-From: Jeff Crews
[mailto:[EMAIL PROTECTED] Sent: Friday, April 16, 2004 5:49
PMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Agent Cleanup Time?Previously
"Unable to join queue
'queuename'". I think that I have the queue setup properly in
queues.conf. I am just not defining any members.
Any help or guidance
that you can give would be greatly appreciated.
Thanks,
Robert
Jackson
with inbound DID's? Everything is
setup properly in *, but I am not able to receive inbound calls, through
VoicePulse of course. It was working properly yesterday, and without
changing anything it stopped working.
Thanks in advance,
Robert Jackson
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me know if it is working as well.
Thanks,
Robert
-Original Message-
From: Isaac McDonald [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 2:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse Connect Problems
Robert Jackson wrote:
Just a quick couple
monitor this list.
Isaac
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf
tones? I have had a terrible time getting a hold of anyone over there,
and I need this functionality before I can migrate to * completely
couldn't find anything that helped me either on the wiki or
the list.
Thanks in advance,
Robert Jackson
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!
Sorry for the rabbit hole guys, but if I had not gotten these
suggestions from ya'll I would have been stuck at this point until I
just gave up. (Or decided to shoot the damned thing, whichever came
first.)
Thanks again,
Robert Jackson
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Try this:
exten = _0.,1,Dial(CAPI/xxx:b${EXTEN:1})
The :1 tells it to use everything except the first digit.
Robert Jackson
-Original Message-
From: massimo [mailto:[EMAIL PROTECTED]
Sent: Friday, April 09, 2004 6:59 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ignorepat
on the bugtracker, but I would rather make sure
that I am not just completely dense and not seeing the easy answer. I'm
trying to replicate the issue with NuFone.
CVS from 2004-04-04 stable branch.
Thanks,
Robert Jackson
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(thankyouforcalling)
exten = s,5,Background(mainmenu-prompts)
exten = 1,1,VoicemailMain()
exten = 1,2,GoTo(s,5)
exten = 2,1,Directory
exten = 2,2,Goto(s,5)
exten = i,1,Playback(invalid)
exten = h,1,Hangup
exten = t,1,Hangup
Thanks for your help,
Robert Jackson
-Original Message-
From: Steven
I completely agree. This way you can get the same functionality on
demand instead of automatically.
-Original Message-
From: Duane [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 08, 2004 5:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] res_motv: Request for Comment
Andy
can't find any other way to do it. Your help is
greatly appreciated.
Thanks,
Robert Jackson
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allow=alaw
allow=gsm
2) The scripts have been moved to the /usr/src/asterisk/contrib/scripts/
subdirectory. Once you run the script it will prompt you for the
context, which I have left blank, and the extension.
3) I don't know because I haven't gotten that far.
Hope this helps,
Robert Jackson
There was a question about this earlier. I had a similar problem and
fixed it by specifying the audio protocol to be used in the general
section of the sip.conf.
-Original Message-
From: Altus Snyman [mailto:[EMAIL PROTECTED]
Sent: Monday, April 05, 2004 3:52 AM
To: asterisk
Subject:
Title: Message
I am just an
Asterisk newbie doing a test install. I am using 2 X-Lite clients and
haveconfigured them according to the wiki on voip-info. A warning is
still displayed on the Asterisk server console saying that I should disable
RFC3389 on the client, even after I changed the
87 matches
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