Due to the size of most larger deployments, expecting the phone's built-in
directory to handle this is probably ... unreasonable.
I believe someone mentioned that there was some work to define this sort
of functionality for SIP in the RFC's, but I don't recall for sure.
I guess the problem
cd /usr/src/zaptel
make config
This will install the init scripts to start zaptel when you boot your computer.
cd/usr/asterisk
make config
does the same thing for asterisk. Use asterisk -r to connect.
Dylan.
On 5/3/05, Ben Johnson [EMAIL PROTECTED] wrote:
When I restart my computer, I need
Chris,
If you are looking to run your second line through *, you will need to
run the line from the demarc to an FXO port on your Asterisk machine,
and then run a line from an FXS card to the jack location where you
will be using an analog phone.
You cannot connect an incoming line from your
Anyone experimented with Calling Card support in * Am I wrong in
presuming that if I have one calling card caller call in and want to
complete a call I will use 2 lines (1 for the customers inbound and another
to complete the remote call)??
If you use IAX2 termination for incoming and
On 4/18/05, Nabeel Jafferali [EMAIL PROTECTED] wrote:
Anyone experimented with Calling Card support in * Am I wrong in
presuming that if I have one calling card caller call in and want to
complete a call I will use 2 lines (1 for the customers inbound and
another to complete the
If the time is off by exactly x hours, check the *timezone* in ipmid.cfg.
On 4/14/05, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote:
Does anyone know how Polycom 500s will be able to update their time. My
setup for a time sync with Public domain Time servers is not successful.
James,
Agents and Queues are part of a system known as Automated Call
Distribution (ACD). Incoming calls go into a queue, where they may
receive some form of call treatment (an initial announcement, hold
music, status updates) before they are answered by call agents. Calls
will never ring at an
Have you tried using an ACD queue?
http://voip-info.org/wiki-Asterisk+Agents
http://voip-info.org/wiki-Asterisk+config+agents.conf
http://voip-info.org/wiki-Asterisk+config+queues.conf
Dylan.
On Apr 11, 2005 9:25 AM, Chris [EMAIL PROTECTED] wrote:
I need to make a time loop in the
I have an auto-attendant for day and night. When the [businesshours]
AA runs, it executes exten = s1 through exten = s9, then continues
with exten = s10 in [nightmode], even though they are in different
contexts. This was working fine until I added more 's' extensions in
night mode. When I comment
Has anyone tried loading the PAP2-NA firmware on the PAP2 Vonage model?
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Implementing the dial plan you described is pretty straightforward.
Integration with your existing voice mail system (Meridian Mail?) will
be a lot harder. You may opt to use seperate voice mail systems, but
exchanging messages between systems is currently not possible. If your
current vm
Dave,
That looks very good. Can't wait to get my hands on it! This is extactly
the type of interface that allows you to retain full control, while
making things easier for the occasional user.
Dylan.
Dave Packham wrote:
OK lets start out with this.
Im not a pro GUI designer ?
Now that
Can you also post the Nortel config? What release software are you
running on the Option 11?
Dylan.
Manuel Marín García wrote:
I am interconnecting a T100P with a Meridian Option 11 1.5 Mb PRI card.
Actually Meridian is working with alaw. The problem is that when I dial from
Meridian side to
I have been seeing references to a ncurses based tool called Newt (on
the Bayonne mailing list of all places). What is Newt? Is it part of the
main Asterisk source?
Dylan.
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Hi everybody,
I've been tinkering with a web based interface for Asterisk. I tried to
stick as closely to the current configuration format as possible. The
web interface should help to do things a little easier (sort by
extension, context, do bulk changes).
www.packetbell.com/asterisk
bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel, newt-devel,
openssl096b, openssl-devel
The GSM library/codec that you're missing is normally installed with SOX, I think.
Let me know if you get it working--nice to see * on a Macintosh.
Dylan.
Hi,
I am trying to compile
That would be the case if calls are dropped at random to clear the way
for 911 calls. With some form of access control (NCOS, Calling Search
Space/Partitions, priority levels) you would be able to drop the least
important calls.
BTW, how are trunk restrictions managed right now? How can I
I've been scouring the archives for discussions on this:
Why doesn't Asterisk use system user accounts for each
extension/mailbox? That would add the benefit of encrypted passwords,
logical grouping, unified mail/voice mail accounts (using
/var/spool/mail instead of /var/spool/asterisk). I can
I'm wondering if LDAP might be the more correct thing to use though.
Absolutely!
Reed Wade wrote:
Because you haven't written and contributed that functionality yet.
(smiley face goes here)
That sounds pretty sweet. I'm wondering if LDAP might be the more
correct thing to use though.
-reed
Problem: 911 calls placed through Asterisk are associated with the
physical location of where the CO trunks terminate. This is not really a
problem when all extensions are located in the same building, but when
Asterisk is used in a campus-like or otherwise networked environment, it
can get
Remove the space behind .com, like so http://asterisk.650dialup.com/
Cheers, Dylan.
Uriel Carrasquilla wrote:
For some reason the page cannot be found.
http://asterisk.650dialup.com
what does it do?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Now that I reed it back, I can barely make sense of it myself! Anyway, I
was just thinking out loud, the example wasn't meant to be parsed.
Asterisk would need some lower level changes to parse the extra field
holding the location information, and to apply the routing rules to
substitute the
Dylan VanHerpen wrote:
Now that I reed it back, I can barely make sense of it myself! Anyway,
I was just thinking out loud, the example wasn't meant to be parsed.
Asterisk would need some lower level changes to parse the extra field
holding the location information, and to apply the routing
And now that I *read* it back again, you can tell that English is not my
native language either
Dylan VanHerpen wrote:
Now that I reed it back, I can barely make sense of it myself!
Anyway, I was just thinking out loud, the example wasn't meant to be
parsed. Asterisk would need some lower
Also, it isn't very easy to 'test' either, as the staff at the 911 call
centre won't appreciate your testing, and at least in Australia, it is some
sort of criminal?/illegal offence to call emergency for non-emergency
situations.
Well, for testing purposes 911 could be replaced with any other
Can Zaptel T1/E1 cards also be used with other Linux apps (for instance
as the WAN interface on a Linux based router, or with Bayonne, Vocal)?
Thks, Dylan.
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Hi everyone,
I'd like to use Asterisk to build a phonetree (www.phonetree.com) type
of application, like this:
1. Read a text-based name/phonenumber file.
2. Call every number and play a recorded message.
3. If a beep is detected, replay the message from scratch (to leave
messages on an
Steve wrote:
On Wednesday 11 June 2003 08:08 pm, Dylan VanHerpen wrote:
Hi everyone,
I'd like to use Asterisk to build a phonetree (www.phonetree.com) type
of application, like this:
1. Read a text-based name/phonenumber file.
2. Call every number and play a recorded message.
3. If a beep
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