View source for Linksys spa-3102

[[Category:Hardware]]
[[Category:Tutorials]]
=Linksys SPA-3102=

====Specifications====
The Linksys SPA3102 features the ability to connect standard telephones and fax machines to IP-based data networks with the additional benefit of an integrated connection for legacy telephone network "hop-on, hop-off" applications.

*IP Telephony features
**VoIP Protocols	SIP v2
**Voice Codecs	G.711, G.723.1, G.729a, G.726
**Telephony Interfaces	1 phone (FXS), 1 line (FXO)
**IP Telephony Features	Echo cancellation (G.168)

=Setting up with linuxMCE=

==Configuring the Device==
*Step 1 - Initial setup.
**Connect the SPA-3102 to your internal network (the one connected to the lmce subnet)
**Proceed with initial configuration via the administration page. Once you set passwords and locale settings, you can move to the next step
**Note, you can leave the dhcp and network settings as is. We will not be using the routing functions of this device. You will need to make a note on the address assigned to the device.
**Log in to the LinuxMCE web admin.  Go to Wizard -> Devices -> Phones -> Add Device (bottom of the page).  For device template select '''Generic SIP softphone'''.  Rename the phone to something like '''SPA-3102 Line 1''' and hit update.  Note the extension of the phone.  You will need it later.

*Step 2 - SIP connection settings. 
**Click the ''PSTN Line'' settings.
**Choose 'Admin' and 'Advanced' option in the top left right corner to see all available options.
***Do not change anything other than what you need. There are a lot of options!
*** ''Line Option'' Set to enable.
*** '''Sip Settings'''
**** Make sure the port is set to '''5061'''
*** '''Proxy and registration'''
**** Proxy: the '''ip''' address of your core. 
**** Register: Yes.
**** Make calls without registration: - Not tested but set to yes.
**** Answer calls without registration: Not tested but set to yes.
*** '''Subscriber Information'''
****For ''Display Name'' I put 'Landline'.
****For ''user id'' I put the extension I created for this line.
****For ''password'' I put the password for the extension
****For 'Use Auth Id' I chose no.
***'''Dial Plans'''
****Under '''Dial Plan 2''' add this ''''(<S0:s>)'''
***'''PSTN-To-VoIP Gateway Setup'''
****PSTN-To-VoIP Gateway Enable: '''Yes'''
****PSTN Ring Thru Line 1: '''No'''
****PSTN CID For VoIP CID: '''Yes'''
****PSTN Caller Default DP: '''2'''
*****This is the dial plan we changed earlier.
*****You may need to change some of the other items such as FXO timing, but thats beyond this article for now.
** Click the ''Line 1'' settings.
*** ''Line Option'' Set to enable.
*** '''Sip Settings'''
**** Make sure the port is set to '''5061'''
*** '''Proxy and registration'''
**** Proxy: the '''ip''' address of your core. 
**** Register: Yes.
**** Make calls without registration: - Not tested but set to yes.
**** Answer calls without registration: Not tested but set to yes.
*** '''Subscriber Information'''
****For ''Display Name'' I put 'Line 1'.
****For ''user id'' I put the extension I created for this phone (see Step 1).
****For ''password'' I put the password for the extension (the extension number by default)
****For 'Use Auth Id' I chose no.

==Configuring FreePBX ==
*Step 1
** Log into the webadmin.
** Using the top navigations, goto Advanced -> Configuration -> Phones Setup.
***This will take you to the freePBX panel. Take care to only change designated settings as you can mess things up.
**On the left hand side, choose '''Trunks''' from the menu.
**This will show you the existing trunks (phone lines). Choose '''Add Trunk'''.
**Then choose '''Add Sip Trunk'''.
**Next are the options presented:
***Outbound Caller ID : Caller ID for calls placed out on this trunk.
***Maximum channels: Set this to '1'
***Dial Rules : '''Still working on this part'''. Ideally, it would be used for local non toll calls as well as 911 or any other local services.

**Outgoing Settings

  context: House Line

  allow=all
  context=DID_trunk_2
  dialformat=${EXTEN:1}
  hasexten=no
  hasiax=no
  hassip=yes
  port=5061
  registeriax=no
  registersip=no
  secret=12345678
  trunkname=Custom - telecom
  trunkstyle=customvoip
  username=pstn

**Incoming Settings

  context: pstn

  allow=ulaw&alaw&g723.1&gsm
  context=from-pstn
  dialformat=${EXTEN:1}
  disallow=all
  hasexten=no
  hasiax=no
  hassip=yes
  host=dynamic
  port=5061
  registeriax=no
  registersip=no
  secret=210
  trunkname=Custom - telecom
  trunkstyle=customvoip
  type=friend
  username=pstn

==Routing Calls==
Currently, the call routing in this setup is incorrect. It will route into the linuxMCE telcom system, but it will not have the proper caller ID information. Im currently working on this. Please see update below


===Setup update===
I was able to get linuxMCE to treat the PSTN phone calls the same as ip calls. I can now get the caller ID (Partially in some cases) from the calls as they come in, and I no longer use the work around below. 

To set this up properly, I did the following:
*Created an Inbound Route in the freePBX panel called 'House Line', the same user context as the incoming settings for my PSTN Trunk.
*Duplicated the settings similar to my Broadvoice settings, but there arent many. Whats critical is to make sure at the bottom, you send it to the custom app with this:
  custom-linuxmce,102,1
*Reload the panel and test it out. 
-golgoj4 19-jul-08


===Setting up a 'Ring All' Extension===
 *NOTE* this is a workaround i originally used in my attempts to figure this out. The setup above is the recommended path.

To bring the call in, i experimented with different methods and found some success here. 
*Go to the freePBX main panel.
*On the left side, choose 'Ring Groups'
*On the top right side, choose ''Add Ring Group''
**Group Description : I put '''House Line'''.
**Ring Strategy: '''Ring All'''
**Extension List: Here I added an extension that didnt exist, so it would move on to the bottom. 
**You can experiment with the other settings, but for now we move onto '''Destination if no answer'''.
**Choose the last option (Custom App)  and enter this into the box:
  custom-linuxmce,102,1
*Save and apply changes.
*Test out the phone line.

=Troubleshooting=
This isn't a proper setup by any means, but it will be improved. 
**You can experiment with the different settings, but I recommend reading these links 1st.
***'''http://www.kernelcrash.com/blog/asterisk-pbx-and-the-spa-3102/2007/12/27/'''