Marie.o: Asterisk-LinuxMCE moved to Asterisk-pluto

2010-05-17T16:13:41Z

Asterisk-LinuxMCE moved to Asterisk-pluto

←Older revision	Revision as of 16:13, 17 May 2010
(No difference)

Juliet at 10:55, 16 March 2007

2007-03-16T10:55:28Z

New page

[[Category:Telecom]]
== About the package ==
LinuxMCE uses Digium software PBX called asterisk, and it packages all needed things at once:

* asterisk (the server itself)
* asterisk-addons (mysql support)
* asterisk-sounds (sound files)
* libpri (library for PRI and ZAP)
* zaptel (kernel drivers dor Digium cards)
* bristuff (patch for support some BRI cards)
* iaxyprov (provisioning utility for iaxy)
* chan_capi (channel for support some ISDN cards)
* chan_sccp (channel to suppport Cisco's SCCP phones)
* app_conference (to replace MeetMe)
* spandsp (library for fax support)
* AMP/freepbx (web interface to configure asterisk, not much better then editing conf files)
* several scripts and custom configuration files

At this moment we use asterisk 1.0.9 (almost a year old), the transition to 1.2.x series is desired, below some information about each of them.

== Asterisk 1.0.9 ==
* Location : 10.0.0.150:~dant/asterisk/
* How to build : ./make_package.sh
* When to rebuild : when the LinuxMCE kernel changes
* Where to put : into replacements repository
* Known issues : shouldn't be any

== Asterisk 1.2.7.1 ==
* Location : 10.0.0.150:~dant/asterisk.1_2/
* How to build : ./make_package.sh
* When to rebuild : when the LinuxMCE kernel changes, anytime in needed
* Where to put : into replacements repository when we really want to do the upgrade
* Known issues
** not so clean upgrade (conflicts on several conf files in /etc/asterisk)
** after install need to enable "core" module in freepbx (if not there yet)
** need to add ARI voice mail patch (to automatically login user into web interface when checking voice mail)
** also check that voice mail works and is accessible from ARI, *98 and Telecom_Plugin
** need to test and probably change following scripts
*** sync_pluto2amp.pl (know not to work)
*** create_amp_*.pl (not tested, but need to check)
*** create_pluto_dialplan.pl (should work)
*** create_telecom_defaults.pl (also should work)
*** delete_phoneline.pl
** don't forget dialplan scripts:
*** pluto-sos-gen.pl
*** pluto-sos-neighbor.pl
*** pluto-sos-speak.pl
** and AGI scripts
*** pluto-gethousemode.agi
*** pluto-getusermode.agi
*** pluto-sos.agi

== Other information ==
Not too much to say here, the package installs asterisk and several scripts which will try to keep asterisk running.

We need to rebuild the package any time LinuxMCE kernel changes or other modifications were made. Also don't forget to bump version if you want to upgrade old installation to new release.

The server itself is beyond my understanding, it's running and when it's properly configured it connects all phones together and allows then to call outside the house with configured phoneline.

For more details please check [http://www.digium.com/en/supportcenter/documentation/ asterisk-support] and [http://www.voip-info.org/wiki-Asterisk asterisk-wiki]

== Troubleshooting ==

The first step in any asterisk troubleshooting should be connecting to asterisk console
asterisk -r
then you can issue commands to inspect the situation and also to visualize device status, call progress, routing and other stuff.

Also a good help in identifying the problems are ''tcpdump'' and ''tethereal''. For more info check this [http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP howto].

=== asterisk not running ===
try
ps auxww | grep '/usr/sbin/asterisk'
if the output is empty try to start it with
/etc/init.d/asterisk restart
if doesn't start after than you have dig deeper, check logs

Running asterisk '''IS A MUST''' for the whole thing to work.

=== phone devices do not work ===
* try to connect to asterisk ''asterisk -r'' and issue
sip show peers
iax2 show peers
sccp show devices
and see if all phones are connected to asterisk
* check in Wizard > Devices > Phones that all phone devices have a phone number
* check in Advanced > Configuration > Phones that each extension is properly configured
* if the device that doesn't work is a '''SCCP phone''' please inspect carefully /etc/asterisk/sccp.conf (check for duplicated entries in device and lines sections)
* if the device that doesn't work is the '''SimplePhone'''(Orbiter's Embedded Phone) then check if it's running ''ps auxww | grep SimplePhone'', check it's log maybe some other program was using soundcard and SimplePhone couldn't get access to it.
* if it's a '''hard phone''' try to rerun manually configure script, check device's web page (if exists) or try some other methods to configure it
* if it's a '''soft phone''' please check in product's manual how to configure it.

=== phoneline doesn't work ===
* check in Wizard > Devices > Phone Lines if every information is correct (user name, password, phone number, host)
* check in Advanced > Configuration > Phones setup if trunks, inbound and outbound routes are correct (should be but those are created/deleted by some scripts, so some of them may failed). Also even if I tried to provide correct setup for all supported providers, there may be issues with some of them (from my fault or provider changed something in it's configuration)
* try to connect to asterisk ''asterisk -r'' and issue
sip show registry
iax2 show registry
and see if you are connected to the provider
* SIP and even IAX2 can be blocked by a firewall somewhere in the middle (SIP is more fragile). There is no sure way to tell you if it's firewall, network problem (like frequent loss of packets, the no network situation is easy to find) or provider's fault (well you can detect provider's fault if other provider using same protocol works)

Asterisk-pluto - Revision history

Marie.o: Asterisk-LinuxMCE moved to Asterisk-pluto

Juliet at 10:55, 16 March 2007