The only way I've worked around it thus far is using nfqueue on linux.
I found it by setting up the SPA to log to a syslog server and getting a TP Parse Error: 32 from the SPA in the logs. If you do set this value to 2 or 3 (long enough to wait for caller id lookup), any call with a caller ID name which contains commas (or any other character that should be escaped) will cause FS to send a 400: Bad From: Header back to the SPA with very little to show for it, don't even see it on the fs_cli without cranking sofia debug up. I read some place to set this to 3 for caller ID to work, but I just wanted it to instantly forward and start ringing ext 1001 PSTN Answer Delay: 0 I want incoming VoIP requests to be dialed on the phone line aka: the (xx.) rule Line 1 VoIP Caller DP: 2 (xx.) basically means dial on the phone line of the SPA3102 Dial Plan 2: (xx.) If you want a different extension than 1001 to be dialed change it here Dial Plan 1: (S0) The FreeSwitch password for the user Password: 1234 Subscriber Information Display Name: PSTN Line NOTE: There is a bug in the default configuration of the SPA-3102 and other Linksys devices that sets the RTP Packet Size to.
Diff derv sip definition install#
You have the initial FreeSwitch install SPA3102 DEVICE CONFIGURATION You have the default SPA3102 configuration (Factory defaults) with web access enabled I believe this is a bug in the SPA and CISCO has been at least made aware of it. If your device is unable to register with FreeSWITCH, changing 'Advanced' > 'Voice' > 'Line 1' > 'SIP Transport' to 'TCP' from 'UDP' did the trick for me. Any rule that does not match the dialed number will be dialed on the SPA3102 (aka: not ext 5000, 9999, 1000-1019. This is a minimal how-to configuration on setting up a SPA3102 with FreeSwitch so incoming calls will be forwarded to FreeSwitch extension 1001 and will show the incoming call as FreeSwitch extension 1000.
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