Revision as of 07:58, 15 June 2008 by Langstonius (Talk | contribs) (→Troubleshooting)
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:
- 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. Idealy, it would be used for local non toll calls as well as 911 or any other local services.
- Outgoing Settings
allow=ulaw canreinvite=no context=from-pstn disallow=all dtmfmode=rfc2833 host=192.168.80.233 incominglimit=1 nat=never port=5061 qualify=yes secret=xxx <----extension password type=friend username=xxx <----username on the spa-3102 that you set should be pstn
- Incoming Settings
- This is currently empty.
- Incoming Settings
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.
Setting up a 'Ring All' Extension
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:
- Save and apply changes.
- Test out the phone line.
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.