=> -Original Message-
=> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
=> boun...@lists.digium.com] On Behalf Of Steve Edwards
=> Sent: Tuesday, September 29, 2009 7:32 PM
=> To: Asterisk Users Mailing List - Non-Commercial Discussion
=> Subject: Re: [asterisk-user
Kevin P. Fleming wrote:
> Where would you suggest this note be placed? We've tried to make our
> documentation as clear as possible that the download selector is the
> canonical place to get the proper FFA modules for any given version of
> Asterisk, and the fact that the newer versions of Asterisk
On Tue, Sep 29, 2009 at 3:11 PM, Doug Lytle wrote:
> [directory]
>
> exten => s,1,Wait(1)
> exten => s,n,Directory(sip|sip|eb)
> exten => s,n,Playback(goodbye)
> exten => s,n,Hangup()
> exten => o,1,Goto(incoming,s,1)
I thought the second arg to Directory() was where it would look for
the "o" ext
On Tue, Sep 29, 2009 at 2:48 PM, David Backeberg wrote:
> On Mon, Sep 28, 2009 at 10:08 PM, sean darcy wrote:
>> On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg
>> wrote:
>>> Have you tried using ps2tiff?
>> I looked up ps2tiff. That seems to be a windows program. There is a
>> pstotiff linux
> Steve Edwards wrote:
>>
>> Is the manager or are the agents using disa()?
>>
>> How about:
>>
>> exten = *,n,set(SPYGROUP=ALLOW-SPYING)
>>
>> for the agents and:
>>
>> exten = *,n,chanspy(,g(ALLOW-SPYING))
>>
>> the manager?
On Tue, 29 Se
Death to all sip users!
Paulh
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- "Danny Nicholas" wrote:
> Two questions: 1. do you need an ActionID line?
Danny,
It's *always* considered best practice to have an ActionID line in AMI
commands, so that you can easily differentiate the responses, especially to
asynchronous commands.
--
Jared Smith
Training Manager
Digi
On Tue, Sep 29, 2009 at 08:07:40PM +, Anahi Ludueña wrote:
>
> Hi people, I need to update the voicemail.conf from the UpdateConfig Action
> (AMI).
> The problem is that I executed:
>
> Action: UpdateConfig
> srcFileName: voicemail.conf
> dstFileName: voicemail.conf
> Action-00:append
>
On Tue, Sep 29, 2009 at 08:19:16PM +0200, jonas kellens wrote:
> Through the optware-package I have installed Asterisk on an external
> USB. Further I have a Linksys WRT610N with DD-WRT v24 mega.
>
> I start asterisk with the following command : /opt/sbin/asterisk -c
> I get the following WARNING
Hello listers,
I'm running Asterisk 1.4.26.1 and 1.4-R201993 (SVN) using
Polycom 501's and POTS. The problem I'm experiencing is that when I dial a
call, it takes 1-4 seconds before I hear a ring. I understand that there is
a delay on POTS connectivity, but what's the deal on a
Hi,
I never try it on 1.6 but any information on further compability with
1.6 could be interesting.
This version was developped for the 1.4 version.
If you need informations about installing or configuring this module,
don't hesitate to contact me.
On 09/29/2009 09:21 PM, Rafael Seste wrote:
Steve Edwards wrote:
>>> On Tue, 29 Sep 2009, John Millican wrote:
>>>
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force. I have
the DISA part working with authentication(rather straight forward) bu
Thanks, the result was:
Response: Success
Anahi Ludueña
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 29 Sep 2009 15:16:52 -0500
Subject: Re: [asterisk-users] UpdateConfig
Two questions: 1. do you need an ActionID
line? 2. did you try this
Two questions: 1. do you need an ActionID line? 2. did you try this in a
telnet session so you could see the feedback?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Tuesday, September 29, 2009 3:08 PM
T
Which version of the LDAP schema? I look after the one in 1.6.
Thanks.
On 29/09/2009, John A. Sullivan III wrote:
> On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote:
>> Hi all,
>>
>> I looked on the Internet but I didn't find any good how-to.
>> I would like to integrate a ldap server ( wit
Hi people, I need to update the voicemail.conf from the UpdateConfig Action
(AMI).
The problem is that I executed:
Action: UpdateConfig
srcFileName: voicemail.conf
dstFileName: voicemail.conf
Action-00:append
Cat-00:test
Var-00:exten
Value-00:>999,test
But I don't see the change
I have a user but I need to give that user only kill and disable all
connection cut calls what is the command in the CLIC
--
Bayardo Sánchez García
Web Developer - Internet Portals - Asterisk Support - Windows Server Support
- Proxy Support - Linux Server
E-mail: bayardo.sanc...@gmail.com
Linux U
>> On Tue, 29 Sep 2009, John Millican wrote:
>>
>>> I have a request for remote users to be able to dial through the system
>>> so that the sales managers can barge/chanspy on the sales force. I have
>>> the DISA part working with authentication(rather straight forward) but
>>> what I can not figur
Steve Edwards wrote:
> On Tue, 29 Sep 2009, John Millican wrote:
>
>> I have a request for remote users to be able to dial through the system
>> so that the sales managers can barge/chanspy on the sales force. I have
>> the DISA part working with authentication(rather straight forward) but
>> w
tks for all answers!!!
Antoine,
I will try to do it tomorrow.
just one question. Do you know if it works with asterisk1.6? I'm using
this version and looks like that your friend is using 1.4
On Tue, Sep 29, 2009 at 12:28 PM, Antoine Patte wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
Paul Dugas wrote:
> [attendant]
> ;
> exten => *,1,NoOp(Attendant: Directory)
> exten => *,n,Directory(default,attendant,eb)
> exten => *,n,Goto(s,1)
>
> exten => o,1,NoOp(Zero)
> exten => o,n,Goto(0,1)
>
> exten => a,1,NoOp(Star)
> exten => a,n,Goto(0,1)
>
Work
On Mon, Sep 28, 2009 at 10:08 PM, sean darcy wrote:
> On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg wrote:
>> Have you tried using ps2tiff?
> I looked up ps2tiff. That seems to be a windows program. There is a
> pstotiff linux program, but it seems to be unmaintained, and isn't
> available on
Trevor Peirce wrote:
> A note somewhere would have been nice explaining this. I recently tried
> the Digium Fax to determine if a client should buy some licenses, but
> after seeing it reject invites for T.38 on incoming calls and never try
> to switch for outgoing, I figured it was just broken
Through the optware-package I have installed Asterisk on an external
USB. Further I have a Linksys WRT610N with DD-WRT v24 mega.
I start asterisk with the following command : /opt/sbin/asterisk -c
I get the following WARNING :
r...@dd-wrt:/opt/etc/asterisk# /opt/sbin/asterisk -c
Asterisk 1.4.22.1
I've got a context in my dialplan like so but pressing 0 doesn't seem
to be working. Instead of dropping out to the "o" extension, it's
just returning to the start of the direcotry app. Same with star.
Anyone see where I've gone awry?
[attendant]
;
exten => *,1,NoOp(Attendant: Director
Kevin P. Fleming wrote:
> I am working on getting this situation resolved and should have new
> releases of FFA out at the end of this week, but in the meantime if you
> want to use FFA with T.38 support you'll have to use one of the versions
> of Asterisk listed on the download selector page.
>
On Tue, 29 Sep 2009, John Millican wrote:
> I have a request for remote users to be able to dial through the system
> so that the sales managers can barge/chanspy on the sales force. I have
> the DISA part working with authentication(rather straight forward) but
> what I can not figure out is h
On Tue, 2009-09-29 at 11:23 -0500, Tilghman Lesher wrote:
> On Tuesday 29 September 2009 10:30:37 John A. Sullivan III wrote:
> > Second, I believe we saw a way we could map the Asterisk password to the
> > regular user password (it's been a while so I'm not sure about that) but
> > were concerned
Scott L. Lykens wrote:
> As we are eliminating our PRI soon I am trying to get faxing via T.38 working
> properly. I'm not interested in running Callweaver next to Asterisk just to
> support fax. :/
>
> Any insight into this error would be greatly appreciated.
This is occurring because the cu
On Tuesday 29 September 2009 10:30:37 John A. Sullivan III wrote:
> Second, I believe we saw a way we could map the Asterisk password to the
> regular user password (it's been a while so I'm not sure about that) but
> were concerned about the problems of entering secure passwords from a
> phone key
I believe that this information is at least indirectly in the CDR.
104
106
DLPN_DialPlan1
"Danny Nicholas" <104>
SIP/104-b790d5f8
SIP/106-084585d0
Dial
SIP/106|20|iKkTtwW
97
92
ANSWERED
DOCUMENTATION
1.25E+09
If you subtract the 92 from the 97, you ge
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of David Backeberg
> Sent: Monday, September 21, 2009 10:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] dig
We've been having a strange problem all day where when making outbound
calls, all we get is static on the far end (i.e we can hear, they
can't).
We've restarted asterisk a couple of times to no avail. It now
transpires that it is only mobile numbers that are affected (not all
mobile networks, not
On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote:
> Hi all,
>
> I looked on the Internet but I didn't find any good how-to.
> I would like to integrate a ldap server ( with all users data) with
> asterisk to authenticate SIP users. With this solution I will only
> need to add a user on ldap,
Hi all,
I would like to know if there is a function/setting for extracting the
call setup time in asterisk (1.4 or 1.6).
I need this value for every call processed by asterisk as specified in
(ETSI TR 101 329-1 v3.1.2):
Call set-up time is the time elapsed from the end of the user
interface
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
A realtime ldap driver exist.
He can put the user/peer sip/iax in a ldap directory and configuration
files.
A friend has updated as part of his final study of.
You can find-it there : http://wiki.ouranos.be/doku.php/stage:ldap
Shaun Ruffell wrote:
> On 09/29/2009 06:52 AM, Jerry Geis wrote:
>> A user report that this issue:
>>
>> https://issues.asterisk.org/view.php?id=15429
>>
>>
>> Has resolved their problem with a TDM card.
>>
>> My card is a T1/PRI card. Different module to load.
>> I have the same issue.
>>
>> Does
On 09/29/2009 06:52 AM, Jerry Geis wrote:
> A user report that this issue:
>
> https://issues.asterisk.org/view.php?id=15429
>
>
> Has resolved their problem with a TDM card.
>
> My card is a T1/PRI card. Different module to load.
> I have the same issue.
>
> Does this same problem exist in the PRI
What you could use would be an AMI interface that does a "core show channels
verbose" to get the active call information, then display that as an HTML
table. When the supervisor clicks on the call he/she wants, the AMI
originates a chanspy/barge command as appropriate. 75% of the responders on
th
Hello all,
OS OpenSuSE 10.3
* ver 1.4.26.2
zaptel ver. 1.12
Digium TE122
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force.
I have the DISA part working with authentication(rather straight
forward) but what I c
I have 3 phones, SIP/3, SIP/6 and SIP/9
SIP/3 subscribes on hint on SIP/9
Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it
is steady red. That's correct.
But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is
still flashing.
It keeps flashing until somebo
You could get the Free PERL module Asterisk::Ldap and use it to periodically
update your users from the LDAP server. You could make it a daily cron job
run at midnight so any new LDAP users would be Asterisk users the new
business day and you could also run the module on-demand.
-Original Mes
Hi all,
I looked on the Internet but I didn't find any good how-to.
I would like to integrate a ldap server ( with all users data) with
asterisk to authenticate SIP users. With this solution I will only
need to add a user on ldap, it will not be necessary to add any
special configuration on sip.co
IMO the easiest way to accomplish this would be to do an AMI call to "core
show channels verbose" and pick out the line containing the extension. You
could also pick out the customer number so a record could be made if another
agent was talking to the customer.
-Original Message-
From: as
Maurizio Faccio adinet wrote:
> I own a TDM2400 board, with three FXO modules and one FXS.
> I'am having trouble with analog sip phones, from two different
> equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202),
> sometimes when I am calling someone, then I press flash, and then
I own a TDM2400 board, with three FXO modules and one FXS.
I'am having trouble with analog sip phones, from two different
equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202),
sometimes when I am calling someone, then I press flash, and then call
someone else, both calls stay c
A user report that this issue:
https://issues.asterisk.org/view.php?id=15429
Has resolved their problem with a TDM card.
My card is a T1/PRI card. Different module to load.
I have the same issue.
Does this same problem exist in the PRI code and needs fixed their also?
Has it been fixed? and do
And of course I forgot the most important stuff:
Asterisk version: 1.4.22
DAHDI Linux: 2.2.0.2
DAHDI Tools: 2.2.0
- "Vinícius Fontes" escreveu:
> I have a pretty large setup on one of my customers. Digium TE420B
> (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each
> and 1 X
Hello,
We need help in debugging Music On Hold on our Asterisk 1.6.1.6
From the SIP debug, I see that an extension sends an INVITE of the call
to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but
I don't see in the console any reference to the call being placed on hold.
Whe
I have a pretty large setup on one of my customers. Digium TE420B (with echo
cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank
with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they
are connected to cell phones. Not really related to the issue,
When I say no reliable internet, some days it's good, others it's not, so to
try to push a call over an IAX trunk is going to fail.
For the choice of providers, when it comes to a business with branches
around the country (NZ is small enough) we'd choose one provider.
Subaddress is not limited to
In relation to our CRM-system I'd like to send a query to asterisk who
is extension xxx talking to.
When the operator enters the page with customer data, the crm should
send a query to asterisk, to get the cli of the call the operator is having.
If the number is matching the customers number in
I didn't think of Benny's solution. That would be the way to go as it is a
core asterisk command.
On Tue, Sep 29, 2009 at 6:47 PM, Vieri wrote:
>
> --- On Tue, 9/29/09, Matthew Edmondson wrote:
>
> > If you redirect the channel, the
> > person they're talking to is likely
> > to be dropped.
>
>
--- On Tue, 9/29/09, Benny Amorsen wrote:
> > I'm wondering if someone can share their thoughts on
> how to implement a system that periodically checks active
> channels which have been up for more than X minutes and
> plays/injects a sound file. The idea is to simply warn users
> that they've b
--- On Tue, 9/29/09, Matthew Edmondson wrote:
> If you redirect the channel, the
> person they're talking to is likely
> to be dropped.
Thanks for pointing that out. So it sounds like RedirectChannel() is similar to
Transfer().
> The only way I know of doing this is with a conference
> bridge
Vieri writes:
> Hi,
>
> I'm wondering if someone can share their thoughts on how to implement a
> system that periodically checks active channels which have been up for more
> than X minutes and plays/injects a sound file. The idea is to simply warn
> users that they've been on the phone for q
If you redirect the channel, the person they're talking to is likely
to be dropped.
The only way I know of doing this is with a conference bridge like
meetme. You would have to have both parties in the conference and then
call the 3rd party (your msg) into it.
On Tue, Sep 29, 2009 at 6:05 PM, Vie
Hi,
I'm wondering if someone can share their thoughts on how to implement a system
that periodically checks active channels which have been up for more than X
minutes and plays/injects a sound file. The idea is to simply warn users that
they've been on the phone for quite a while and maybe they
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