Slimming Asterisk for the NSLU2 under Debian

Slimming Asterisk for the NSLU2 under Debian

A FULLY WORKING ASTERISK SIP PBX RUNNING WITH ONLY 6 MODULES LOADED ! READ ON…

This howto is based on Asterisk 1.2 under Debian Etch. Please let me know through the comments if it works for you under other versions (and if it doesn’t, please provide the steps to get a working system). Thanks.

My needs :

I had to slim Asterisk down to the most minimalistic configuration possible to run on my Linksys NSLU2 (ARM cpu @ 266 Mhz, RAM 32 MB).

- SIP calls between my IP phones and softphone- Incoming/Outgoing calls through a SIP ITSP (ipness.com)- Echo test to make sure audio is going through in some situation- I only use the alaw codec, compatible with my IP phones and ITSP, you should avoid transcoding. My DSL connection offers dynamic IP only and gives around 3400 Kbps down/386 Kbps up. I could use the GSM codec, but the ITSP doesn’t support it- No voicemail or other apps

My setup :

- The NSLU2 is behind a NAT router- Home phone on the same subnet as the NSLU2- Work phone behind NAT- Softphone used from several places

Router configuration :

- Forward port UDP/5060 to UDP/5070 to the NSLU2- UDP/5060 is used for SIP traffic (signalling)- UDP/5061 to UDP/5070 is used for RTP traffic (audio)

Asterisk configuration files :

Before diving into the configuration..IMPORTANT !!!If you want to comment something out in the configuration you’ll start the line with a semi-colon (“;”)# is used for file inclusions. # IS NOT USED FOR COMMENTS !!!

I moved unnecessary files under backup/

​​root@nslu2:/etc/asterisk# ls -ltotal 40-rw-r--r-- 1 root root 247 2008-04-13 22:03 asterisk.confdrwxr-xr-x 2 root root 4096 2008-04-13 22:03 backup-rw-r--r-- 1 root root 141 2008-04-13 22:10 extensions.conf-rw-r--r-- 1 root root 1660 2008-04-13 23:34 features.conf-rw-r--r-- 1 root root 2158 2008-04-13 22:03 logger.conf-rw-r--r-- 1 root root 438 2008-04-13 23:37 modules.conf-rw-r--r-- 1 root root 395 2008-04-13 22:03 rtp.conf-rw-r--r-- 1 root root 299 2008-04-13 22:04 sip.conf-rw-r--r-- 1 root root 904 2008-04-13 23:40 custom_extensions.conf-rw-r--r-- 1 root root 1349 2008-04-14 09:31 custom_sip.conf​​

/etc/asterisk/extensions.conf :

[general]static=yeswriteprotect=noautofallthrough=yesclearglobalvars=nopriorityjumping=no; let's include custom_extensions.conf into extensions.conf#include "/etc/asterisk/custom_extensions.conf"

/etc/asterisk/features.conf : default configuration

/etc/asterisk/logger.conf : default configuration

/etc/asterisk/modules.conf :

Order in which modules are loaded can be important. E.g. : res_features.so must be loaded before chan_sip.so

[modules]autoload=no ; only load explicitely declared modulesload => app_echo.so ; echo applicationload => codec_alaw.so ; alaw codec for voiceload => pbx_config.so ; reading and loading configurationload => res_features.so ; chan_sip.so dependencyload => chan_sip.so ; SIP protocolload => app_dial.so ; Dial application[global]

/etc/asterisk/rtp.conf :

The audio is going through these UDP ports, they must be forwarded to the server in the router.

In this example, the number of ports used by Asterisk is relatively low (I never have more than one call going through the PBX). Set as much as you need, a channel may need up to 2 ports. Also only even ports are actually used.

​​[general]rtpstart=5061rtpend=5070​​

/etc/asterisk/sip.conf :

Having a dynamic IP, I must use externhost with a fresh rate of 60 seconds for resolving the domain.If you have a static IP, define it in externip=

Localnet must be defined with your network subnet(s), localnet subnets are never passed in the “Via” parameter (can be seen in sip traces).

realm must be a unique ID

The “register =>” line is necessary in order to receive incoming calls from the ITSP.

[general]context=incomingbindport=5060bindaddr=0.0.0.0srvlookup=yesexternhost=mydynamichostname.dyndns.orgexternrefresh=60localnet=192.168.1.0/255.255.255.0realm=mydynamichostname.dyndns.orgregister => 3221234567:password:login@ipness-:6060/3221234567#include "/etc/asterisk/custom_sip.conf"

/etc/asterisk/custom_extensions.conf :

custom_extensions.conf is my customized dialplan

- Home phone and softphone can call local phones (1XXX range)- Home phone and softphone can call national (e.g. : 02 123 45 67) and international numbers (00 1 910 123 4567) through the ITSP- Work phone can only call local phones- Every phone can call the echo application- Incoming calls make 1001 ring first, then 1002 and finally 1000 (each with a timeout of 30 seconds)

[globals];;; apps context[apps]exten => 444,1,Answer()exten => 444,n,Wait(1)exten => 444,n,Echo;;; incoming calls context[incoming]exten => 3221234567,1,Dial(SIP/1001,30)exten => 3221234567,n,Dial(SIP/1002,30)exten => 3221234567,n,Dial(SIP/1000,30);;; outgoing calls context; local calls only[local]include => appsexten => _1XXX,1,Dial(SIP/${EXTEN})exten => _1XXX,n,NoOp(===== DIAL STATUS --> ${DIALSTATUS} =====)exten => _1XXX,n,Hangup(); national (Belgium, code 32) calls only[national]include => localinclude => appsexten => _0N.,1,Dial(SIP/0032${EXTEN:1}@itsp_ipness)exten => _0N.,n,NoOp(===== DIAL STATUS --> ${DIALSTATUS} =====)exten => _0N.,n,Hangup(); international calls[international]include => nationalinclude => localinclude => appsexten => _00.,1,Dial(SIP/${EXTEN}@itsp_ipness)exten => _00.,n,NoOp(===== DIAL STATUS --> ${DIALSTATUS} =====)exten => _00.,n,Hangup()

/etc/asterisk/custom_sip.conf :

custom_sip.conf is my customized accounts file.

canreinvite must be set to no for all sip accounts (unless you have several phones on the server subnet, than you can set to yes).NAT must be set to yes for any device behind NAT routers.

; SIP accounts[1000]type=friendcontext=internationalcallerid="Softphone" <1000>qualify=yessecret=1111nat=yescanreinvite=nodtmfmode=rfc2833host=dynamiccall-limit=2disallow=allallow=alaw[1001]type=friendcontext=internationalcallerid="Home" <1001>qualify=yessecret=1111nat=no ; IP phone is on the same subnet as the servercanreinvite=nodtmfmode=rfc2833host=dynamiccall-limit=2disallow=allallow=alaw[1002]type=friendcontext=localcallerid="Work" <1002>qualify=yessecret=1111nat=yescanreinvite=nodtmfmode=rfc2833host=dynamiccall-limit=2disallow=allallow=alaw[1003]type=friendcontext=localcallerid="Friend" <1003>qualify=nosecret=1111nat=yescanreinvite=nodtmfmode=rfc2833host=dynamiccall-limit=1disallow=allallow=alaw; ITSP[itsp_ipness]type=peerusername=loginsecret=passwordfromuser=3221234567fromdomain=ipness-host=ipness-port=6060nat=yescanreinvite=nocontext=incominginsecure=veryqualify=yesdisallow=allallow=alaw

Result :

nslu2*CLI> show modulesModule Description Use Countapp_echo.so Simple Echo Application 0codec_alaw.so A-law Coder/Decoder 0pbx_config.so Text Extension Configuration 0res_features.so Call Features Resource 1chan_sip.so Session Initiation Protocol (SIP) 0app_dial.so Dialing Application 0 nslu2*CLI> show applications -= Registered Asterisk Applications =- AbsoluteTimeout: Set absolute maximum time of call Answer: Answer a channel if ringing BackGround: Play a file while awaiting extension Busy: Indicate the Busy condition Congestion: Indicate the Congestion condition Dial: Place a call and connect to the current channel DigitTimeout: Set maximum timeout between digits Echo: Echo audio read back to the user ExecIfTime: Conditional application execution based on the current time Goto: Jump to a particular priority, extension, or context GotoIf: Conditional goto GotoIfTime: Conditional Goto based on the current time Hangup: Hang up the calling channel ImportVar: Import a variable from a channel into a new variable NoOp: Do Nothing Park: Park yourself ParkedCall: Answer a parked call Progress: Indicate progress ResetCDR: Resets the Call Data Record ResponseTimeout: Set maximum timeout awaiting response RetryDial: Place a call, retrying on failure allowing optional exit extension. Ringing: Indicate ringing tone SayAlpha: Say Alpha SayDigits: Say Digits SayNumber: Say Number SayPhonetic: Say Phonetic Set: Set channel variable(s) or function value(s) SetAccount: Set the CDR Account Code SetAMAFlags: Set the AMA Flags SetGlobalVar: Set a global variable to a given value SetLanguage: Set the channel's preferred language SetVar: Set channel variable(s) SIPAddHeader: Add a SIP header to the outbound call SIPDtmfMode: Change the dtmfmode for a SIP call SIPGetHeader: Get a SIP header from an incoming call Wait: Waits for some time WaitExten: Waits for an extension to be entered -= 37 Applications Registered =-nslu2*CLI> show functionsInstalled Custom Functions:--------------------------------------------------------------------------------CHECKSIPDOMAIN CHECKSIPDOMAIN() Checks if domain is a local domainSIPCHANINFO SIPCHANINFO(item) Gets the specified SIP parameter from the current channelSIPPEER SIPPEER([:item]) Gets SIP peer informationSIP_HEADER SIP_HEADER() Gets the specified SIP header4 custom functions installed.

Asterisk is using around 12 MB of memory when idle.

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