[asterisk-dev] Number of digit to trunk??

Francois dr.viza at vizatelecom.net
Fri Jan 4 08:35:56 CST 2008

Hello every body.

I am using asterisk . I have Mediant 2000 as voip gateway. In my
country the short number ( I mean police, hostpital etc.. ) uses only 2
digit number for example 01, 02, 03 , 09 etc.but my problem is that when I
dial the 2 digits number Asterisk doesn't send it to Gateway. It just wait
for few second and gives message that I have dialed the wrong number. I have
checked on Gateway Log file, no message from Asterisk about short number.
Can anybody tells me where to change the minimal authorize number of digits
for so that Asterisk can send it to Gateway?




From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Eugene Grossi
Sent: Friday, January 04, 2008 3:42 PM
To: 'Raj Jain'; 'Asterisk Developers Mailing List'
Subject: Re: [asterisk-dev] Implementation of Broadsoft Sip Access in
Asterisk to enable SLA for Sipura/Linksys


In my patching, I have only dealt with (enabled) a single (first) line


Eugene A. Grossi, M.D.
Professor of Cardiothoracic Surgery
New York University School of Medicine
Suite 9-V; 530 First Avenue
New York, NY 10016
ph (212) 263-7452
fax(212) 263-5534
grossi at cv.med.nyu.edu


From: Raj Jain [mailto:rj2807 at gmail.com] 
Sent: Friday, January 04, 2008 5:07 AM
To: Asterisk Developers Mailing List
Cc: grossi at cv.med.nyu.edu
Subject: Re: [asterisk-dev] Implementation of Broadsoft Sip Access in
Asterisk to enable SLA for Sipura/Linksys



Somewhere in my mail archive I have a permission from Broadsoft or 
Cylantro to implement this - before they
published it as a draft. They wanted Asterisk to include this
functionality. After that, they published it openly
as a draft, so there should be no problems.

It's probably also supported by the fact that this specification is publicly
available on BroadSoft's company website: 

If we speak about SLA specifically, they've basically extended SIP on three

1. Additional parameters in the existing Call-Info header 
2. Call-Info SIP event-package
3. Line-Seize SIP event-package

The Call-Info event-package pretty much accomplishes the same thing as the
standard dialog event-package (i.e. it notifies the state changes of the
shared line to SLA stations). 

The Line-Seize event-package solves an arbitration issue that happens in
en-bloc signaling protocols such as SIP. It allows the entire SLA group to
be immediately informed as soon as one SLA station seizes the shared line.
This allows an SLA station to "win" the line. The other phones render the
line busy while one phone is dialing. I've seen different commercial PBX
vendors solve this problem in their own proprietary way. 

There is one more interesting aspect to these SIP extensions -- they're
built on the concept of Call Appearance. Call Appearance means that a unique
line can have multiple calls active on it at the same time. So, you're not
only telling the phones which line underwent a state change, but also which
appearance of it. 

Does the current Asterisk SLA implementation support the Call Appearance


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