, with xen 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.
Robe
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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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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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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luck and welcome,
Robert Jackson
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plicated. I forgot the KISS method ;)
FYI - (I just suggested that he use ${EXTEN:-8})
Robert Jackson
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from ${EXTEN} 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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As
ys,
but in the meantime I would do:
# cvs checkout -D 01/24/2005 asterisk asterisk-addons
This will get you 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
tool, and many topics have
already been discussed.
Good luck,
Robert Jackson
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uggestions I'm looking for
> wholesale pricing for termination as we are a CLEC.
>
Please do not cross 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
on as we are a CLEC.
>
> Any suggestions would be greatly appreciated!
>
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
__
oySigurd Karlsbakk)
>
>
> There:
> https://sourceforge.net/tracker/index.php?
> func=detail&aid=746083&group_id=29880&atid=541465
> Added IAX ping :)
>
> roy
>
Thank you very much. That is exactly what I was looking for
as wel
his every day.
Check out the following 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
o see a function created to do this.
Please let us know what you find or end up doing.
Robert Jackson
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I believe that *2
activates the recording, but I could be wrong.
Robert Jackson
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The
system should stop at the first match.
Good luck,
Robert Jackson
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d+CheckGroup
Also, Please try not to use HTML e-mail on the list. ;)
Hope this helps,
Robert Jackson
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fo.org/tiki-print.php?page=Asterisk+auto-dial+out
Hope this helps,
Robert Jackson
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wing context:
[acd-context]
exten => 24XX,1,SetGroup(${EXTEN})
exten => 24XX,2,CheckGroup(1)
exten => 24XX,3,Dial(SIP/${EXTEN})
exten => 24XX,103,Busy
This will also keep the ACD calls from going to a persons voicemail box,
which would probably happen if your queue member didn'
ingall strategy and
have the devices as static members.
Good luck!
Robert Jackson
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ll dial extension 1234, or if they hit 1 it
will drop to voicemail. You will have to test it to see what happens
when the caller hits a button not defined.
If you do test it please let us know how it behaves when an extension
is entered, but not sp
* http://www.voip-info.org/wiki-Asterisk+config+extensions.conf -
Dialplan Intro
* http://www.asteriskdocs.org/ - Asterisk Documentation project
* http://www.voip-info.org/ - Asterisk Wiki
Hope this helps, and good luck.
Robert Jackson
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ackLogin. This way you can have one set of
behaviors for reaching an agent at an extension and another set for
simply reaching the extension outside of an ACD context.
This is how we have it setup and it seems to work pretty well.
Hope this helps,
Robert Jackson
___
e called in put something like:
exten => _1XXX,1,Ringing
exten => _1XXX,2,Dial(SIP/${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 musi
regarding multiple links
If that isn't your goal please disregard.
Good luck,
Robert Jackson
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the strategy
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,
s.
Since * uses the current context for transfers this can be a
problem. The solution to all of this was to
SetVar(TRANSFER_CONTEXT=) before
you call Dial(,20,t). Then * uses the context that
you specified to use for transfers.
Fixed a similar problem for us,
Robert Jackson
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, but I am not sure.
This message is rather annoying so I would definitely like to
see if anyone else has gotten it figured out.
Robert Jackson
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us, and it
allows the person recieving the transfer to let it go to
voicemail or choose to answer it.
(Also, posted to the bug tracker.)
Hope this helps,
Robert Jackson
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> -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-
>
t;exten => _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
> >exten => _91NXXNXX,2,Macro(dial-result)
> >
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-st
keys that the key is loaded and listed exactly like I have
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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ht
.0. I want the latest
> w/ bug fixes but no new features.
>
cvs checkout -r v1-0 will get you the latest for version 1.0
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
___
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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a in regards to the last notification sent 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
y, it selected the records periodically
(I think it was every 360 seconds) and added them to the linked
list which made the mwi work.
Just my $.02 worth,
Robert Jackson
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utomatically log them
off if they don't answer. This is a bit more complex
especially if you have many remote extensions.
Just a couple of ideas,
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-
To my knowledge
there is no way to write a file directly from within the dialplan.
Hope this helps,
Robert Jackson
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normal context execute the macro
instead of Dial.
Integrating these two features together has allowed 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 sa
g more code in the check_availability.
I am also going to take a look at #3 while I have things opened up.
Please let me know if we are on the same page.
Robert Jackson
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same perl script which loops
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 prett
=> 2688,3,Hangup
>
I had a similar problem with my system, and I 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,Mee
two files together for me.
Hope this helps,
Robert Jackson
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onf
* http://www.voip-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")
H
> something stable that doesn't crash.
>
Those commands certainly do download the latest up-to-the-minute
Version of asterisk. It is possible that some bugs and whatnot
will be found, but I have found that CVS Head is pretty stable.
Untill RC1 was relea
ta, but I am not
quite there yet.
Also, if you aren't populating from a database do you have
[EMAIL PROTECTED] in your voicemail.conf?
Hope this helps,
Robert Jackson
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in. I believe that it is what you are
looking for.
http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin
Good luck,
Robert Jackson
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roblem with CVS 8/4/2004.
Thanks for your help,
Robert Jackson
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re,
> will investigate further.
>
I believe that it is for use with IAX, but I could be wrong.
Robert Jackson
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auto attendant where 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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eing it?
>
I think that it should 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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ceMail(b${CALLEDEXTEN})
exten => 1,3,Hangup
I found this in the bug notes of bug # 509.
http://bugs.digium.com/bug_view_page.php?bug_id=509
Hope this helps,
Robert Jackson
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outbound calls. I ended up disabling call waitingfor those agents, and
not allowing the calls to go to voicemail.
Thanks,
Robert Jackson
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To UNSUB
We are running a 20 agent call center using *. We are using the
built-in Queues/Agent support. We haven't really had any major issues,
other than my own configuration mistakes. I certainly does everything
that we need it to.
Hope this helps,
Robert Jackson
-Original Message-
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]
>
erything is good depending 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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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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After just having update to the latest CVS I am getting the following
message when I call VoicemailMain():
-- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack
-- Playing 'vm-login' (language 'en')
-- Playing 'vm-password' (language 'en')
-- Playing 'vm-youhave' (language '
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
> Subj
ou answering the line first? I 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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ember 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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[EMAIL PROTE
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.
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 Hansel
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
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.
> -Origi
strative_q]
music=default
announce-holdtime=yes
announce-frequency=90
strategy=ringall
;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
me
all calls with an exten => _. 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
-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 just
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
$sth->fetchrow_array();
my $ssn = $data[0];
my $dob = $data[1];
my $dob = $data[1];
my $zip = $data[2];
my $homePhone = $data[3];
$AGI->verbose("Data returned: $ssn, $dob, $zip, $homePhone",3)
} else {
$AGI->verbose("I
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
l throw an error "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
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 bef
in and
let 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 qu
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
__
!
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
___
Ast
sterisk-Users] Newbie Issues => SIP won't stay connected,
and IAX Unable to Create Channel
Robert Jackson wrote:
> I am terribly sorry to bother the list with such generic and bizarre
> problems, but I have been racking my brain with these for the last
> week working on it for a
loss. Again I apologize for wasting your valuable
time, but I couldn't find anything that helped me either on the wiki or
the list.
Thanks in advance,
Robert Jackson
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gt; s,4,Background(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,
Rober
n 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
___
Asterisk-Users ma
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] Ignore
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 Pow
can't find any other way to do it. Your help is
greatly appreciated.
Thanks,
Robert Jackson
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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: [As
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,
Robe
Title: Message
I am just an
Asterisk newbie doing a test install. I am using 2 X-Lite clients and
have configured 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 th
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