Revision as of 07:34, 10 June 2008 by Langstonius
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.
- 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 '(<S0:600>)
- 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.
- 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: