What about pattern matching / QueueMembers? Does that work for you? If
it does please let me know since I might be doing something wrong
Regarding password/email - that IS the part that goes into LDAP
(password, email,options). If that doesnt work there is no point to
using LDAP Voicemail
unsuscribe
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Hello Sebastien
First this is not the way to unsubscribe
Second you can do better with your spelling too.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastien
Mortier
Sent: Sunday, March 16, 2008 4:30 PM
To: Asterisk Users Mailing List -
Cavalier did not leave me any paperwork to sign. I check with the colo as
well and they did not receive anything. I have a feeling I have not defined
my trunk groups well in zaptel.conf.
On 3/15/08, Ron Joffe [EMAIL PROTECTED] wrote:
On Saturday 15 March 2008 21:22, Darren Wright wrote:
If you write a HowTo, would you please insert it into the wiki at
http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks.
On Sun, 2008-03-16 at 07:09 -0500,
[EMAIL PROTECTED] wrote:
Date: Sat, 15 Mar 2008 18:20:32 -0200
From: Gonzalo Servat [EMAIL PROTECTED]
Subject: Re:
James Finstrom wrote:
Anyone have the telemarketer torture prompts? I would seriously like
to revive this.
Weasels and Monkeys work well for this.
I put up one extension that uses Monkeysintro then Monkeys and loops.
The other extension uses somethingwrong then weasels and again loops
back
On Sunday 16 March 2008 08:09, broadband Voice wrote:
Cavalier did not leave me any paperwork to sign. I check with the colo as
well and they did not receive anything. I have a feeling I have not defined
my trunk groups well in zaptel.conf.
Your zapata.conf looks ok.
Try this for your
On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese [EMAIL PROTECTED] wrote:
I just forward them to one of those two extensions. If callerid worked
more reliably I would automate it. But I get a lot of caller id failures
on my incoming POTS lines, esp when calling in from my cell phone.
The way I
James Finstrom wrote:
Anyone have the telemarketer torture prompts? I would seriously like to
revive this.
I created a simple hold forever loop, thus:
[tele_loop]
exten = s,1,Answer()
exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE)
exten = s,3,WaitMusicOnHold(15)
exten = s,4,Wait(1)
I have simply gotten rid of most of the telemarketer calls by simply
making them press 1 or wait to talk to me and they give up almost
always.
on Sunday 03/16/2008 Horwich IT Services (Godwin Stewart)([EMAIL PROTECTED])
wrote
On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese [EMAIL PROTECTED]
I used to deal easily with telemarketer calls way before I ever got
into asterisk. No hardware, no dial plan, no coding, no recording and
less disk-thrashing than any of the previous systems. I'd just let
them get started, set the phone down and continue what I was doing
before they called.
/r
James Finstrom wrote:
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Anyone have the telemarketer torture prompts? I would seriously like
to revive this.
- --
I wrote one a while back that uses Cepstral TTS, but the mechanics are simple.
When a telemarketer calls, I say hmmm, that
That is silly. Why are you locked into their entire IDS, disro, settings.?
Thanks,
Steve Totaro
On Fri, Mar 14, 2008 at 4:34 PM, Joshua Wilson [EMAIL PROTECTED] wrote:
You need to download and install the system from the iso image. This puts
everything into place on the system.
On
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I realize there are ways of doing it but I am kinda fond of the loops
etc of the telemarketer torture...
For those who are unaware:
http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture
Lee Jenkins wrote:
James Finstrom wrote:
I'm looking for a usb cordless handset to pair with a softphone (
probably ekiga) on a pc linked to an asterisk server. I've loooked at
the bluetooth headsets, but they seem overkill for just home phone
extensions.
I've found a number of handsets that work with skype, but they seem
locked
James Finstrom wrote:
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I realize there are ways of doing it but I am kinda fond of the loops
etc of the telemarketer torture...
Same here, I want to keep them on the line as long as possible, thinking
that they may eventually get to someone.
You could accept as the passcode the caller punching in their own
phone#, then checking that against your whitelist. Lets associates get
past the challenge when using someone else's phone, without their
remembering some arbitrary passcode.
And strangers or barred old associates
It is the same whether you are using trixbox, switchbox, pbxware or any
other system roughly. You have to use their system. Sometimes, this is the
only way to make sure everything they support is installed and integrated
properly without problems.
On 3/16/08, Steve Totaro [EMAIL PROTECTED] wrote:
On Sun, Mar 16, 2008 at 12:36:12PM -0600, Joshua Wilson wrote:
It is the same whether you are using trixbox, switchbox, pbxware or any
other system roughly. You have to use their system. Sometimes, this is the
only way to make sure everything they support is installed and integrated
properly
Joshua Wilson wrote:
It is the same whether you are using trixbox, switchbox, pbxware or any
other system roughly. You have to use their system. Sometimes, this is
the only way to make sure everything they support is installed and
integrated properly without problems.
Hi Joshua,
PBXware
On Sun, Mar 16, 2008 at 3:27 PM, Senad Jordanovic [EMAIL PROTECTED] wrote:
Joshua Wilson wrote:
It is the same whether you are using trixbox, switchbox, pbxware or any
other system roughly. You have to use their system. Sometimes, this is
the only way to make sure everything they support
Hi List,
What is the best way to implement this solution with several branches?
SIP PHONE 1 --- *SERVER 1 ---WAN--- *SERVER 2 --- SIP PHONE 2
WAN
SIP PHONE 3 --- *SERVER 3 ---WAN--- *SERVER 4 --- SIP PHONE 4
The Asterisk boxes will have one leg on the LAN side and
Steve Totaro wrote:
On Sun, Mar 16, 2008 at 3:27 PM, Senad Jordanovic [EMAIL PROTECTED] wrote:
Joshua Wilson wrote:
It is the same whether you are using trixbox, switchbox, pbxware or any
other system roughly. You have to use their system. Sometimes, this is
the only way to make sure
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Vieri wrote:
--- Ex Vito [EMAIL PROTECTED] wrote:
...as long as the destination does not answer
you'll get
a NO ANSWER disposition.
So, if in your case you want to know if a user
answered
the phone, then, yes, you will have to add the
Hi all,
I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
Broadvoice TOO often, however I have a Vermont number with them and so
my mother in law calls it to talk to my wife once in a while, so
that's why it took me so long to notice it wasn't working. Anyway,
when she calls she
iax.conf doesn't take canreinvite=no . It's only for sip . For iax2 its
transfer=yes/no (1.4 ) or notransfer=yes (1.2) , there's one more parameter
in 1.4 with which u can transfer only audio stream . Check voip-info wiki
for all options .
On Thu, Mar 13, 2008 at 7:48 AM, Gonzalo Servat [EMAIL
Looking at the trace, the entity sending you the INVITE is not
resubmitting INVITE with credentials after the initial INVITE was
challenged with a 401 response by Asterisk. The trace shows two
independent calls and both have the same problem.
--
Raj Jain
mailto:rj2807 at gmail dot com
sip:rjain
well, maybe ou're on the wrong list (talkin sendmail in an asterisk list
!!!) you're better in sendmail's list.
anyway, you need to modify sendmail.cf file, just a few tweaks it will be
ok. you will need a smarthost, what is a smarthost ? thats an smtp server
that is allowed to send mail to the
I am wanting to call a shell script in System() command and it does not
seem to work.
If I put a direct linux command it does work.
My script is owned by root and is executable.
Is this possible? What might I be missing?
Jerry
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I tried that and got 14 errors, see below:
[EMAIL PROTECTED] etc]# service zaptel restart
Unloading zaptel hardware drivers:ERROR: Module zaptel is in use
.
Loading zaptel framework: [ OK ]
Waiting for zap to come online...OK
Loading zaptel hardware modules:
Hi Raj,
Thanks for your response.
I'm a little confused though. Does this look as if it's a problem
with Broadvoice itself, and not my configuration? Any time I've
called them with problems where it's clearly not my fault (ie nothing
on my end has changed), they're never very helpful.
On Sun,
Based on the trace alone, it seems like a problem on their end. You
may want to try shutting off INVITE authentication (by commenting out
secret= line in your sip.conf) to see if the call goes through.
On Sun, Mar 16, 2008 at 6:27 PM, Jon Miron [EMAIL PROTECTED] wrote:
Hi Raj,
Thanks for
Great job on the new site...
i found some really great people to do some asterisk installs that i needed to
have done for clients through your site hope your new site does well! i'll
be using your site for anything i have in the future for sure.
--
Matt #1
The 330/550/650 phones have a built-in 2-port switch that speaks
802.1q. Usual use of this is to send two VLANs down the wire. The
phone is configured to use one, and the phone transparently passes the
other to the phone's PC port. On Cisco, this would be a trunk port
with two VLANs, one for the
On Sun, 16 Mar 2008, Jerry Geis wrote:
I am wanting to call a shell script in System() command and it does not
seem to work.
If I put a direct linux command it does work.
My script is owned by root and is executable.
Is this possible? What might I be missing?
A path?
Try specifying the
On Sunday 16 March 2008 18:19, broadband Voice wrote:
I tried that and got 14 errors, see below:
Sorry, switch those around, I gave you a zapata.conf, your original
zaptel.conf looks fine.
Ron
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I am writing an extension to accept speed dial nos.
However, I forgot that these speed dials are the same for all offices
and thus would ideally be shared by offices which will host their own
Asterisk box.
I read from a few postings that this database cannot be replicated to
other Asterisk box.
I
Forgive me if this has been covered before. I did search but I was
unable to find a reference.
I am curious to know more about the possibility of using SQL to store
voicemail as well as having more than one voicemail system accessing a
central SQL database. Any information would be appreciated.
Lee, John (Sydney) wrote:
I was thinking that if I could just do a simple copy/paste of the speed
dial records from the main database to others would be good.
You'd be better off going with a SQL setup. I have ours setup with 1
master and 2 slaves. The slaves are the remote facilities.
You could place all the information in a MySQL database, and either
reference that or replicate from that.
PaulH
On Mon, 2008-03-17 at 11:46 +1100, Lee, John (Sydney) wrote:
I am writing an extension to accept speed dial nos.
However, I forgot that these speed dials are the same for all
On Sun, Mar 16, 2008 at 4:00 PM, Senad Jordanovic [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
On Sun, Mar 16, 2008 at 3:27 PM, Senad Jordanovic [EMAIL PROTECTED]
wrote:
Joshua Wilson wrote:
It is the same whether you are using trixbox, switchbox, pbxware or any
other system
I may be wrong (wouldn't be the first and hopefully very far from the
last) but this just *screams* Shill. I probably wouldn't even buy
it on a client testimonials page.
http://en.wikipedia.org/wiki/Shill
A shill is an associate of a person selling goods or services or a
political group, who
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