[Asterisk-Users] Queue falls through to personal voicemail
This seems like a simple configuration but I must be missing something. I have a queue setup with three members, if I do ring-all every phone gets one ring then one of the three continues to ring and the other two show missed calls. After a few seconds the call goes to that persons voicemail box. If I set it to round robin it will call the next available agent and put the call into their voicemail if they don't answer. Am I missing something or doesn't it seem like it should try the next available agent instead of giving up and going to voicemail? Thanks, Paul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sporadic voicemail delete problems
I have an asterisk server that specific users have delete=yes set in the voicemail.conf file. They are occasionally still recieving the voicemails in their voicemail inbox as well as their email inbox. Is this a known issue, if so is there any work around? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail 0 for operator call routing
Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 - voicemail - Secretary 1 555-1235 - voicemail - Secretary 2 Any help would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 0 for operator call routing
Thats what I was afraid of ;) I was hoping someone would have some magical solution that was easier like an undocumented feature of voicemail that would let me set the operator extension... Never hurts to ask. Don Pobanz wrote: Paul Tinsley wrote: Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 - voicemail - Secretary 1 555-1235 - voicemail - Secretary 2 You will need to set up two contexts for your phones. If someone dials someone who should reach secretary 1 when they 0 out of voicemail, then send them to context1. For secretary 2, send them to context2. [desks] exten = 5551234,1,GoTo(context1,5551234,1) exten = 5551235,1,GoTo(context2,5551235,1) [context1] exten = 5551234,3,Dial(Zap/10,18,t) exten = 5551234,4,Voicemail,us5551234 exten = 5551234,104,Voicemail,b5551234 exten = o,1,dial(secretary1) [context2] exten = 5551235,3,Dial(Zap/11,18,t) exten = 5551235,4,Voicemail,us5551235 exten = 5551235,104,Voicemail,b5551235 exten = o,1,dial(secretary2) Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B
yrving rivas wrote:
Hi all:
I just bought and installed a TDM04B (4 fxo ports). It is running and ok.
I have 2 lines from my provider.
I can make two incoming calls come through my box to its extensions,
but only one going out.
I would like to make 2 calls from different extensions at the same
time from my box out.
¿Where can I find information about? (I couldn´t find any at
http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki)
I´ll appreciate your help.
Regards,
Yrving
Do You Yahoo!? La mejor conexión a Internet y *2GB* extra a tu correo
por $100 al mes. http://net.yahoo.com.mx
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Could you please post your extensions.conf section that references how
outbound calls work? My assumption would be that you are only
referencing one of the zap interfaces and should either be using a group
or listing the zap interfaces sequentially in the dialplan so that if
the first one is in use the second one will be used.
Example of using a group would be, in zapata.conf for the lines you are
trying to dial out on:
group=2
in extensions.conf:
TRUNK=ZAP/g2
exten = _NXX,1,Dial(${TRUNK}/${EXTEN})
exten = _NXX,2,Congestion
Doing it without a group would work as follows:
exten = _NXX,1,Dial(Zap/1/${EXTEN})
exten = _NXX,2,Dial(Zap/2/${EXTEN})
exten = _NXX,3,Congestion
Using a group lets you setup several lines to be used when the group is
referenced.
More complex than you probably need but the letter used when referencing
the group will change how the group works:
; g: select the lowest-numbered non-busy Zap channel
;(aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
;(aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last
;time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
;time (aka. descending rotary hunt group).
So Zap/g1 starts at the lowest line and Zap/G1 starts at the top of the
available lines, etc
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Re: [Asterisk-Users] PERL AGI DIALSTATUS
While his if statement will never do what he expects (check for the
value ANSWER in $dialstatus) it should work when the call is answered
as an assignment in an if statement will pretty much always return
true... and he isn't overwriting a variable he uses in his execute
so the thing should work in that case.
What you mean for your code to say is this:
$AGI-exec('Dial', $dialext);
my $dialstatus = $AGI-get_variable(DIALSTATUS);
if($dialstatus eq ANSWER) {
$Accounting_update-execute($fdatetime,$Cuniq,$UserName,$CalledN);
}
But like I am saying above, that won't fix your problem, without your
other code there is no way to really diagnose the problem. The only
other thing I notice from the blurb is that you have a capital A in
accounting where every other variable you reference other than $AGI is
in lowercase. Are you using strict to make sure you are referencing
variables that exist? You can do that by adding the following line to
the top of your script:
use strict;
you might also try changing the execute line to say:
$Accounting_update-execute($fdatetime,$Cuniq,$UserName,$CalledN) or
print STDERR update failed!;
So that you would know if it failed
Furqan wrote:
Hi Code Lover,
Please try this code :
if ($dialstatus eq
ANSWER){$Accounting_update-execute($fdatetime,$Cuniq,$UserName,$CalledN);}
I hope it will solve your problem. We use eq instead of = in perl
or cgi for strings.
Thanks
Furqan Ahmed
Software Engineer
B.E. (Computer System), DBA
Super Technologies Inc., Pensacola, Florida
Http://www.SuperTec.com - Tommrow's Technology, Today.
Http://www.SuperTec.com/solutions/ - VOIP Billing Solutions.
Http://www.AsteriskSupport.com - Premium Asterisk Support
Development Services.
- Original Message - From: Code Lover [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, December 18, 2005 2:13 AM
Subject: [Asterisk-Users] PERL AGI DIALSTATUS
Hi all,
I wanted to execute one of mySQL query when the call is answered i
tried with the following code but it dones not seems to work.
$AGI-exec('Dial', $dialext);
my $dialstatus = $AGI-get_variable(DIALSTATUS);
if($dialstatus=ANSWER){$Accounting_update-execute($fdatetime,$Cuniq,$UserName,$CalledN);}
It is not updating my query when the call is successfull answered, and
i checked my query from outside using perl commond it is working well
without any issue.
--
Thank You,
Code Lover
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Re: [Asterisk-Users] SRV Lookups
I fail to understand your combative attitude towards an open source project... Did you pay to download it? Are you having to pay to use it? If you really want to talk about throwing money around to fix your problems the bounty idea has already been thrown at you, but also keep in mind that Digium has a custom development service that you could pay for... I'll even give you the link for free! http://www.digium.com/index.php?menu=service_categorycategory=development Douglas Garstang wrote: I guess we can put that up there with the inability to share a common Realtime database between Asterisk servers for SIP peers too... another serious limitation. -Original Message- From: Douglas Garstang Sent: Sunday, December 11, 2005 12:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SRV Lookups Sounds like your saying that a serious limitation that effectively makes Asterisk unusable in a production environment isn't a priority for the 'official' developers. Awesome... -Original Message- From: Leif Madsen [mailto:[EMAIL PROTECTED] Sent: Sunday, December 11, 2005 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SRV Lookups On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote: Anyone know when Asterisk is going to properly support DNS SRV Lookups? Well, luckily Asterisk is open source so you have the ability to code this yourself. If you can't program in C (like myself), then you have the option of either hiring someone directly. Another option is to create a bounty and see if anyone else also requires this functionality and is willing to contribute some money for development. Leif Madsen http://www.oreilly.com/catalog/asterisk http://www.leifmadsen.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
