Asterisk - the open source soft PBX (MIPSel) - Revision history

Ramuk: Articles/MIPSelAsterisk moved to Asterisk - the open source soft PBX (MIPSel)

Ramuk — Sun, 23 Jul 2006 02:08:57 GMT

Articles/MIPSelAsterisk moved to Asterisk - the open source soft PBX (MIPSel)

← Older revision

Revision as of 02:08, 23 July 2006

Ramuk at 01:09, 21 July 2006

Ramuk — Fri, 21 Jul 2006 01:09:27 GMT

← Older revision		Revision as of 01:09, 21 July 2006
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-	<center><font color=red>''Originally by Scotty from linkstationwiki.org''</font></center>	+	{{Template:Articles}}<center><font color=red>''Originally by Scotty from linkstationwiki.org''</font></center>

	== Building/Running Asterisk on Linkstation ==		== Building/Running Asterisk on Linkstation ==

Bauldrick at 17:19, 29 June 2006

Bauldrick — Thu, 29 Jun 2006 17:19:29 GMT

← Older revision		Revision as of 17:19, 29 June 2006
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		+	<center><font color=red>''Originally by Scotty from linkstationwiki.org''</font></center>
		+
	== Building/Running Asterisk on Linkstation ==		== Building/Running Asterisk on Linkstation ==

Ramuk at 03:53, 28 June 2006

Ramuk — Wed, 28 Jun 2006 03:53:32 GMT

← Older revision		Revision as of 03:53, 28 June 2006
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	[[Category:LS2]]		[[Category:LS2]]
		+	[[Category:Howto]]

Ramuk at 23:22, 25 June 2006

Ramuk — Sun, 25 Jun 2006 23:22:10 GMT

← Older revision		Revision as of 23:22, 25 June 2006
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-
	== Building/Running Asterisk on Linkstation ==		== Building/Running Asterisk on Linkstation ==

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	Oh and those limitations - well I couldn't figure out how to build the GSM codec (see http://linkstationwiki.org/forum/3_672_0.html) - so that causes problems when you try to get Asterisk to play any IVR type prompts to the caller. This is not a show stopper, and I think you can rerecord the prompts in PCM (ALAW) and it should work - although I haven't tried it. It's a bit of a shame, cos Asterisk comes with a load of prompts out the box...oh well.		Oh and those limitations - well I couldn't figure out how to build the GSM codec (see http://linkstationwiki.org/forum/3_672_0.html) - so that causes problems when you try to get Asterisk to play any IVR type prompts to the caller. This is not a show stopper, and I think you can rerecord the prompts in PCM (ALAW) and it should work - although I haven't tried it. It's a bit of a shame, cos Asterisk comes with a load of prompts out the box...oh well.
		+
		+	[[Category:LS2]]

Bauldrick: /* modules.conf */

Bauldrick — Thu, 15 Jun 2006 09:53:00 GMT

modules.conf

← Older revision		Revision as of 09:53, 15 June 2006
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	Now you can run it...just type:		Now you can run it...just type:

-	/usr/sbin/asterisk -cvvvvvvv	+	/usr/sbin/asterisk -cvvvvvvv

	...there you have it - Asterisk is now running and configured for 2 SIP extensions 1000 and 1001 that can dial each other. Now you need some SIP phones to register with it and off you go - I'll let you find those. To do some of the more advanced stuff in Asterisk, you will need to specifically load the necessary modules in modules.conf - I have deliberately only loaded those 'important' (to me!) ones.		...there you have it - Asterisk is now running and configured for 2 SIP extensions 1000 and 1001 that can dial each other. Now you need some SIP phones to register with it and off you go - I'll let you find those. To do some of the more advanced stuff in Asterisk, you will need to specifically load the necessary modules in modules.conf - I have deliberately only loaded those 'important' (to me!) ones.

Bauldrick: /* modules.conf */

Bauldrick — Thu, 15 Jun 2006 00:21:08 GMT

modules.conf

← Older revision		Revision as of 00:21, 15 June 2006
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	[modules]		[modules]
	autoload=no		autoload=no
-		+
	load => res_features.so ; Call Parking Resource		load => res_features.so ; Call Parking Resource
	load => res_indications.so ; Indications Configuration		load => res_indications.so ; Indications Configuration

Bauldrick: /* extensions.conf */

Bauldrick — Thu, 15 Jun 2006 00:20:53 GMT

extensions.conf

← Older revision		Revision as of 00:20, 15 June 2006
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	[default]		[default]
	[from-sip]		[from-sip]
-		+
	exten => _[1-9].,1,Dial(SIP/${EXTEN})		exten => _[1-9].,1,Dial(SIP/${EXTEN})

Bauldrick: /* sip.conf */

Bauldrick — Thu, 15 Jun 2006 00:20:42 GMT

sip.conf

← Older revision		Revision as of 00:20, 15 June 2006
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	dtmfmode=rfc2833		dtmfmode=rfc2833
	context=from-sip		context=from-sip
-		+
	[1000]		[1000]
	type=friend		type=friend
	secrect=password		secrect=password
	host=dynamic		host=dynamic
-		+
	[1001]		[1001]
	type=friend		type=friend
	secret=password		secret=password
	host=dynamic		host=dynamic
-

	=== extensions.conf ===		=== extensions.conf ===

Bauldrick at 00:19, 15 June 2006

Bauldrick — Thu, 15 Jun 2006 00:19:48 GMT

New page

== Building/Running Asterisk on Linkstation ==

This article describes how I built and installed Asterisk - the open source soft PBX - on a Linkstation. I am a Linux newbie - so please forgive me. I have a MIPS LSII and I've installed the 052b Openlink firmware as described elsewhere. Previously I have installed MySQL/Apache and PHP as described on this site. If you've done that - then the somewhat terse description below will make perfect sense. As a disclaimer, this little lot took me over a day to figure out - and I haven't been back and retraced my steps - so I appologise if some of the following contains errors - but if nothing else, it'll save you some time.

== Job1 ==

Create a symbolic link in /usr/bin to /bin/awk (because the Asterisk build system expects it there - although I guess you could also change the build system instead)
cd /usr/bin
ln -s /bin/awk awk

== Job2 ==

Download and build the libcurl libraries (Asterisk will use them)

'''* Download the tar.gz from http://curl.netmirror.org/libcurl/'''

'''* Unpack it somewhere sensible'''

'''* Do the usual configure/make/make install (I used the same scheme proposed in the Apache and MySQL articles and installed it into /mnt/hda/opt/libcurl using the --prefix parameter on ./configure)'''

'''* Once built, create a symbolic link to curl-config in /bin'''

'''* cd /bin'''

'''* ln -s /mnt/hda/opt/libcurl/bin/curl-config curl-config'''

'''* Copy the headers and libraries to the 'proper' place'''

'''* copy the /mnt/hda/opt/libcurl/include/curl directory to /usr/include (resulting in /usr/include/curl)'''

'''* copy the contents of /mnt/hda/opt/libcurl/lib to /usr/lib'''

'''* NB: Don't know if there isn't a more elegant way to add the curl include/lib directories to the 'path' for compilation (in Windows you would simply update the INCLUDE and LIB environment vars!)'''

== Job3 ==

Download the Asterisk source (1.4.2 at time of writing) from http://www.asterisk.org

'''* No need for Zaptel and Libpri stuff (you'll not fit a telephony card into a Linkstation ;-)'''

'''* Unpack it somewhere sensible (/mnt/hda/share/asterisk)'''

'''* Next there are a couple of modules that I just couldn't get to build - loads of assembler stuff - so don't build them - this results in some limitations detailed at the end - basically we are just not building the GSM and LPC10 codecs. Edit the code makefile (/mnt/hda/share/asterisk/codec/makefile) and comment out the following (add # to start of line)'''

#$(LIBGSM):
# $(MAKE) -C gsm lib/libgsm.a
#$(LIBLPC10):
# $(MAKE) -C lpc10 all
#codec_gsm.so: codec_gsm.o $(LIBGSMT)
# $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(LIBGSM)
#codec_lpc10.so: codec_lpc10.o $(LIBLPC10)
# $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(LIBLPC10) -lm

'''* Now just hit make and wait for an hour...(notice how we don't need configure)'''

'''* cd /mnt/hda/share/asterisk'''

'''* make'''

'''* Assuming it builds ok - you then need to install it'''

'''* make install'''

== Configuring Asterisk ==

Now it's all built, you can configure it up for a couple of SIP phones. You are now entering the realm of Asterisk config - so I recommend you take a look at a few of the following links when you get into trouble.

http://www.voip-info.org
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.html
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/internals/docs-html/book1.html
http://www.asteriskguru.com/tutorials/

Anyway, we need to create 3 config files in /etc/asterisk as follows:

=== sip.conf ===

[general]
context=default
disallow=all
allow=alaw
dtmfmode=rfc2833
context=from-sip

[1000]
type=friend
secrect=password
host=dynamic

[1001]
type=friend
secret=password
host=dynamic

=== extensions.conf ===

[general]
[globals]
[default]
[from-sip]

exten => _[1-9].,1,Dial(SIP/${EXTEN})

=== modules.conf ===

[modules]
autoload=no

load => res_features.so ; Call Parking Resource
load => res_indications.so ; Indications Configuration
load => res_musiconhold.so ; Music On Hold Resource
load => pbx_config.so ; Text Extension Configuration Requires N/A
load => pbx_functions.so ; Builtin dialplan functions - Requires N/A
load => pbx_loopback.so ; Loopback Dialplan Switch - Requires N/A
load => pbx_realtime.so ; Realtime Dialplan Switch - Requires N/A
load => pbx_spool.so ; Outgoing Spool Support Requires - N/A
load => func_callerid.so ; Gets or sets Caller*ID data on the channel. - Requires ?
load => func_enum.so ; ENUMLOOKUP and TXTCIDNAME functions - Requres ?
load => func_uri.so ; URI encode/decode functions - Requires ?
load => chan_features.so ; Provides summary information on feature channels- Requires N/A
load => chan_local.so ; Show status of local channels- Requires N/A
load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so
load => codec_alaw.so ; A-law Coder/Decoder - Requires N/A
load => format_pcm_alaw.so ; Raw aLaw 8khz PCM Audio support - Requires N/A
load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so
load => app_softhangup.so ; Hangs up the requested channel - Requires N/A

Now you can run it...just type:

/usr/sbin/asterisk -cvvvvvvv

...there you have it - Asterisk is now running and configured for 2 SIP extensions 1000 and 1001 that can dial each other. Now you need some SIP phones to register with it and off you go - I'll let you find those. To do some of the more advanced stuff in Asterisk, you will need to specifically load the necessary modules in modules.conf - I have deliberately only loaded those 'important' (to me!) ones.

You should be able to hook up your Linkstation PBX to an internet telephony provider and use it like a 'proper' PBX - it's got voicemail and IVR functionality etc. That's for another article.

Oh and those limitations - well I couldn't figure out how to build the GSM codec (see http://linkstationwiki.org/forum/3_672_0.html) - so that causes problems when you try to get Asterisk to play any IVR type prompts to the caller. This is not a show stopper, and I think you can rerecord the prompts in PCM (ALAW) and it should work - although I haven't tried it. It's a bit of a shame, cos Asterisk comes with a load of prompts out the box...oh well.