Displaying 6 results from an estimated 6 matches for "2nd_edition".
2009 Aug 20
1
Call routing between two Asterisk boxes using SIP not working ...
Hello there!
I need some help to configure two Asterix boxes to route calls using SIP.
I followed the instructions present at this site:
"http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html",
but I couldn't get it working so far.
The only difference, besides the names that I've used, is that I'm using
realtime to retrieve all information.
Both boxes registrate on the other perfectly.
The problem happens when on...
2013 May 10
1
ISP trunk session ID?
Hi folks,
What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html
My provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my POTS number to my PBX. I ran into couple of bumps on the road but now it's "half-working". I extracted the SIP user, pass, ser...
2010 Nov 15
4
Best way to connect to a MySQL Database
Is this command the best way to access a MySQL database -
MYSQL(Connect connid dhhost dbuser dbpass dbname) ?
I thought I heard that using ODBC was a bit more stable.
Anyone have any experience?
Thanks,
Matt
2009 Oct 14
2
Queues with unavailable members
We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP registration for those unfortunately
doesn't go away just because the phone is off, since the registration is
done by our cell-phone<=>SIP gateway, and that gateway has no way of
knowing whether the phone is on or off.
This is usually ok, but it gets problematic if the cell phone is a
2013 Jan 14
1
php programming for working with asterisk
Hi,
I write some php code in AMI to working with asterisk command. I don't know
exactly what is the different between AMI and AGI and witch one is better
for my planning.
Im planning to call party users that their number is is my panel on web.
We have some operator and they can call party users via client softphone by
clicking on their number, so they have to limited to call just listed
2009 Jul 08
10
q: install asterisk + asteris-gui
hi, i
@asterisk
- svn-ed asterisk from digium 1.6
- make install
>> its running and i can access the CLI
@gui
then i
-svned asterisk-gui from digium
- installed
- repointes apache /var/www/1234 >> /var/lib/asterisk/static_html
>> now, i see the login box, but i dont have any credentials. tutorials are
suggestion manager.conf, BUT I DONT HAVE that file, in fact /etc/asterisk is