Displaying 20 results from an estimated 40000 matches similar to: "[SPAM] [asterisk-dev] CALL FOR COMMENTS - Dialplan"
2007 Apr 20
1
Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemailto text translation)
(This subthread is more appropriate to -users than to -dev, so it is
crossposted only to mark its transition. Please reply on the -user list
only.)
What are the cheapest prices for (humans) transcribing voicemail to
text as a service? The absolute cheapest, regardless of (known) quality
- the quality only has to compete with (cheaper) automated
transcription, which is abysmal quality.
On Wed,
2006 Mar 15
3
Zaptel compile errors on x86_64
Hi,
Just downloaded the latest cvs from zaptel on my sparking new Athlon64
Centos4.2 system, but hitting a stumbling block... (sorry for the long post)
#make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64
2006 Oct 30
1
Anyone got a dialplan for SPA ATAs for ISN?
After John Todd's talk at Astricon about the ISN project, I spent much
of the weekend playing around with it.
I have discovered that the default dialplans on my Sipura gear, as well
as my Grandstream phones, intercept the "*" key that is a required part
of ISN numbers and interpret it as a "metacharacter."
Googling for a while has turned up evidence that this can be
2007 Feb 01
0
Dialplan programming vs. AGI vs. ???
This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that
2006 Feb 22
2
Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.
Hello Asterisk community.
We have a small User-group in Melbourne Australia.
Recently I brought up the issue of STANDARDS for dialing Applications on
a PBX.
This generated some interest but also the fact little has been done on
this topic.
Below is a rundown of our THREAD. (start from bottom and go up)
I myself, feel this to be an important issue. With Asterisk being so
programmable,
2006 Oct 14
1
Re: Generate Random Numbers in dialplan
On Sat, 2006-10-14 at 12:00 -0700,
asterisk-users-request@lists.digium.com wrote:
> Steve,
>
> Is RAND available in the latest trunk or do I need the 1.4
> beta?
>
> If I do show function RAND it says its not available.
>
> Thanks,
> Jon
Jon--
Forgive me, you didn't say which version you
2006 Dec 20
0
Re: Match a Numer - then continue with, dialplan
I seriously doubt he'd know how to get on the 'Internets'
-----Original Message-----
From: Doug Crompton [mailto:doug@crompton.com]
Sent: Wed 12/20/2006 8:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
On Wed, 20 Dec 2006, Michael Collins wrote:
> After listing all of that,
2006 Dec 19
1
SPAM-LOW: Re: .Call files do not seem to wo rk
If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document >
File Format > Unix Format.
I ran into this same problem, and it turns out my Asterisk install would not
use Windows-formatted text files, it would just ignore them
2006 Jun 12
1
TDM-400 and dialplan -- how to ring a SIP ex tension *before* answering the PSTN line?
the caller is out his money anyway when you call any phone and voicemail
kicks in, although i think on a payphone they give you a 2 or 3 second
window to hang up.
Suggest you implement i'm here / i'm away dialplan logic or set the do not
disturb button that way when someone calls and the guy is away it hits
voicemail right away and the caller can hear this and still have the 2 or 3
2007 Jan 17
0
Re: [asterisk-dev] Question about FXO/FXS device.
Okay, i'll move my discuss to asterisk-users.
Thank you.
On 1/17/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote:
>
>
> On Wed, Jan 17, 2007 at 04:39:03PM +0800, ??? wrote:
> > Jonson Player wrote:
> > > Hello, I intend to buy a FXO/FXS device from Linksys.
> > > I'm thinking about SPA3102. What you guys thik about it.
> > > Is ok, is
2007 Apr 04
0
Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemail to text translation)
(This subthread is more appropriate to -users than to -dev, so it is
crossposted only to mark its transition. Please reply on the -user list
only.)
What are the cheapest prices for (humans) transcribing voicemail to
text as a service? The absolute cheapest, regardless of (known) quality
- the quality only has to compete with (cheaper) automated
transcription, which is abysmal quality.
On Wed,
2006 Jan 30
1
Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk
It can be done.
1. Setup a new Vm profile on CCM with a mask of XXXX
2. Setup a CTI route point:
a. Set the directory number to a pattern. I use *27XX
but any pattern that you can send from * is good, ie. 88XXX
b. Set the VM profile to the newly created profile
c. Set the line to forward all calls to VM
3. Change the dialplan in * to append the extension called to
the
2008 Dec 05
3
Rate My Dialplan Contest Announced - Win a Phone or Copies of APSTel Visual Dialplan Std or Pro!
We often find ourselves reading through all sorts of contests on the
Internet that never seem to echo our own personal skill set or interests.
Perhaps you've even fantasized about a type of contest with the types of
prizes and goodies that YOU'D actually enjoy. Maybe you've wished there were
something along the lines of a asterisk phone system diagram contest? With
prizes ranging from
2006 Dec 20
3
Re: Match a Numer - then continue with, dialplan
> -----Original Message-----
> From: David Gomillion [mailto:dgomillion@eyecarenow.com]
> Sent: Wednesday, December 20, 2006 10:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: Match a Numer - then continue with,
> dialplan
>
>
> I think you're making it far too difficult.
>
> What I do is something like this:
>
> [outgoing]
2006 Apr 28
1
[SPAM] [asterisk-dev] Disable 407 proxy authentication for outbound domains
Hello,
I posted a lot of mails may be asterisk is not able to
accept sip calls from internet !?
My english is not fluent i try my best !
My problem I use ser+asterisk.
For local calls there are no problem (PSTN or IP)
Now i wish to receive calls from other internet domain
but asterisk ask for authentication 407.
IS IT possible to Disable authentication for incoming
calls to my sip uri ?
2006 May 01
0
Spam? Re: CallerID Name problem
I'm getting Number but when I look at the CDR database. I do see the name
-----Original Message-----
From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com]
Sent: Mon May 01 17:10:26 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Spam? Re: [Asterisk-Users] CallerID Name problem
Do you get caller ID number? If so, WAITing is not going to help, since you
2006 Dec 05
0
[Fwd: RE: any possibility of Vonage Integration]
I stand corrected!
However you do get my point ...
The bigger the company the worse it is. Having to deal with these guys is
a nightmare. The company that brings me out in spots is Rogers Cable
(24/7). They have this electronic air-head called "Gertrude" or something,
(an android) who can't understand the word "NO" and has trouble with "YES"
(actually like my
2006 Dec 03
0
VoIP GSM Gateways
Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills the same way as he answers followups ;-)
g
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com on behalf of Peter Bowyer
Sent: Sun 03-Dec-06 8:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP GSM Gateways
Not very good at answering followups to
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir,
How did you set sip:tzafrir@local.xorcom.com
I use ser----asterisk
look at my sip.conf and extensions.conf
Regards
Harry
////////////////////////////////////////////////////
[general]
context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
2005 Sep 13
0
PRI zap channels not cleared when nomatchincontext for dialed number on inbound call
Yeah the "variable stays there" because the channel is never up to be
cleared. If you do something like
exten => _X.,1,Wait(1)
exten => _X.,2,Hangup
You will see the same behavior. Can you confirm??
I am running CVS from about a week ago...
Alex
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
>