Difference between revisions of "Asterisk-pluto"

Latest revision as of 17:13, 17 May 2010

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 asterisk-support and 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 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)