[asterisk-bugs] [Asterisk-GUI 0015027]: Asterisk GUI and Dialplan configuration

Mon May 18 17:32:16 CDT 2009

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15027 
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Reported By:                fonto
Assigned To:                awk
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Project:                    Asterisk-GUI
Issue ID:                   15027
Category:                   General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk GUI Version:       SVN 
Asterisk Version:           1.6.0.6 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2009-05-04 08:24 CDT
Last Modified:              2009-05-18 17:32 CDT
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Summary:                    Asterisk GUI and Dialplan configuration
Description: 
Hello, 

First, I'm sorry for my English but I am French.

I try to create a dial plan with the asterisk-gui and teh Outgoing Calling
Rules section.

When it is finished, I look at the extensions.conf file, everything is ok
but priorities are all at 1 and I would like them to be 1,2,3... or n ((as
all normal dialplan).

I try to understand how it works by looking at the callingrules.js file
but I can't program in JavaScript.

I posted some pictures to illustrate the problem: 
http://olivierf66.free.fr/image/asterisk-extension.JPG
http://olivierf66.free.fr/image/asterisk-gui.JPG

P.S: Asterisk 1.6.0.6 and asterisk-gui SVN-branch-2.0-r4759
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 (0104981) Romik (reporter) - 2009-05-18 17:32
 https://issues.asterisk.org/view.php?id=15027#c104981 
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That's not bug, it's a feature.
That is what Outgoing ruleas are _n_ot designed to. They are simply for
adding sone general rules like "if dield number is not out local, then mach
it against appropriate trunk and dialout via it".

To create rules like you want, you should use "Voice menus".

Additionally, there is a need of proper "HOWTO/FAQ" section on the left
panelof Asterisk-GUI with the contents like^
1. Login with user and password you've configured (as it was recommended
in `make checkconfig`) in /etc/asterisk/manager.conf

2. Go to System Status and see that... That that's your system :)

3. Go to Trunks and create some of them, which ever you have: SIP, IAX,
E1/T1 channels, analog lines.

4. Go to Outgoing Calling rules and define that "for numbers that fit
certain pattern we should use this one trunk as first and that one as
backup". For an example, E1 trunk we named "to_our_telco_via_city_via_E1"
on step 3. we use for _XXXXXXX pattern which maches any 7-digit number
which is for our calls to the city and call that "to_city_via_E1". And
another SIP trink we named "to_sip_provider_providername" on step 3. for
all numbers that mach pattern _8XXXXXXXXXX and call it
"to_longdistance_via_sip_provider_providername" (starts from 8 and 11
digits long), create another one _810XXXXXXXXXXXX. and name it
"to_international_via_sip_provider_providername" (starts from 810 and
_at_least_ 15 digits long) => via "to_sip_provider_providername", and as
soon as it is cheaper to call local regional mobiles via E1 city trunk,
than via "to_sip_provider_providername", create another several outgoing
rules with patterns _89172XXXXXX, _89061XXXXXX
("to_mobile_1_via_city_via_E1", "to_mobile_2_via_city_via_E1") which will
use "to_city_via_E1" trunk. Note that _8XXXXXXXXXX pattern overlaps
_89172XXXXXX, _89061XXXXXX, so we should move _89172XXXXXX, _89061XXXXXX
up, othervise we will not get what we want. The rules that are more hungry
should be at the bottom, and rules that are less hungry (more precise)
should be at the top of the list.
Also note that here are only those rules that are to be used to dial from
your local office to the outside world.

5. Go to Dial Plans and define several rules like:
- we have managers that are allowed to call anywhere - put crosses in all
boxes and Save
- we have team members that are allowed to dial there, there and there.
Check appropriate boxes and Save.

6. Go to Users and create several users and assign them to Dial Plans
you've created on step 5. Manger's phones will use different dialplan than
others, etc. Check that phone accounts you've created are GREEN in the
System Status section. That means your phones successfully registered on
the server.

7. Skip at this moment all advanced features like Ring Groups, Music On
Hold, Call Queues, procceed to Voice Menus. Create one.
[add an example here]

8. Skip time intervals for now. They are to make a choice basing on week
day and time which Incoming Call Rule to use: one for work-hours and
another for holidays.

9. Incoming Calling Rules are used for all calls that came to your system
_from_ Trunks. Create one using Voice Menu you have created on step 7.
[add an example here]

10. Click Apply Changes on the top left corner of your browser's window.

11. Test your system. Login to your Asterisk server via SSH and run
`asterisk -rvvvv` from root. Check from newly added and registered in Users
section SIP-phone how can you call to another SIP-phone you've registered
in the same section, check if you can dial to your mobile (call should go
via "to_city_via_E1" trunk), call from mobile to your external numbers that
are coming to your E1 and you will hear meny you've created on step 9.
Relax.

It's 2.31AM here in Russia, I'm going to have a sleep. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-18 17:32 Romik          Note Added: 0104981                          
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