6 months back, I ordered number of Polycom 330 phones + Linksys SPA-3102+ Netgear Switch of 24 Gig PoE ports. During the construction, I planned for each room a simple one cable cat6.
By this, I’m ready to have my IP telephone service ready for implementation. Trixbox is my choice as it’s running FreePBX. I used a very nice manual written by one of Trixbox fans called “Trixbox without tears”
The box is amazing, if you read it and follow the instructions step by step, surely you will get your telephony running in 2 hours. I faced some issues related to Linksys SPA since what was written in the book related to Australia, where we have in SAUDI ARABIA different parameters for the PSTN ( the fixed telephony service from STC ).
Below is the complete configuration for my SPA if any in saudi arabia would like to reuse it. Note : it is not perfect. still I’m facing some echo issues.
| Product Information | |||
| Product Name: | SPA-3102 | Serial Number: | FM600J123931 |
| Software Version: | 3.3.6(GW) | Hardware Version: | 1.4.5(a) |
| MAC Address: | 000E08C0A285 | Client Certificate: | Installed |
| Customization: | Open | ||
| System Status | |||
| Current Time: | 3/12/2010 08:35:44 | Elapsed Time: | 5 days and 07:02:39 |
| RTP Packets Sent: | 1571568 | RTP Bytes Sent: | 251450560 |
| RTP Packets Recv: | 1399891 | RTP Bytes Recv: | 223976008 |
| SIP Messages Sent: | 31654 | SIP Bytes Sent: | 16552671 |
| SIP Messages Recv: | 31553 | SIP Bytes Recv: | 16841323 |
| External IP: | |||
| Line 1 Status | |||
| Hook State: | On | Registration State: | Registered |
| Last Registration At: | 3/12/2010 08:35:24 | Next Registration In: | 38 s |
| Message Waiting: | No | Call Back Active: | No |
| Last Called Number: | Last Caller Number: | ||
| Mapped SIP Port: | |||
| Call 1 State: | Idle | Call 2 State: | Idle |
| Call 1 Tone: | None | Call 2 Tone: | None |
| Call 1 Encoder: | Call 2 Encoder: | ||
| Call 1 Decoder: | Call 2 Decoder: | ||
| Call 1 FAX: | Call 2 FAX: | ||
| Call 1 Type: | Call 2 Type: | ||
| Call 1 Remote Hold: | Call 2 Remote Hold: | ||
| Call 1 Callback: | Call 2 Callback: | ||
| Call 1 Peer Name: | Call 2 Peer Name: | ||
| Call 1 Peer Phone: | Call 2 Peer Phone: | ||
| Call 1 Duration: | Call 2 Duration: | ||
| Call 1 Packets Sent: | Call 2 Packets Sent: | ||
| Call 1 Packets Recv: | Call 2 Packets Recv: | ||
| Call 1 Bytes Sent: | Call 2 Bytes Sent: | ||
| Call 1 Bytes Recv: | Call 2 Bytes Recv: | ||
| Call 1 Decode Latency: | Call 2 Decode Latency: | ||
| Call 1 Jitter: | Call 2 Jitter: | ||
| Call 1 Round Trip Delay: | Call 2 Round Trip Delay: | ||
| Call 1 Packets Lost: | Call 2 Packets Lost: | ||
| Call 1 Packet Error: | Call 2 Packet Error: | ||
| Call 1 Mapped RTP Port: | Call 2 Mapped RTP Port: | ||
| PSTN Line Status | |||
| Hook State: | On | Line Voltage: | -50 (V) |
| Loop Current: | 0.0 (mA) | Registration State: | Not Registered |
| Last Registration At: | Next Registration In: | ||
| Last Called VoIP Number: | s@192.168.15.215 | Last Called PSTN Number: | 017860008 |
| Last VoIP Caller: | Last PSTN Caller: | , 503159152 | |
| Last PSTN Disconnect Reason: | VoIP Call Ended | PSTN Activity Timer: | 60000 (ms) |
| Mapped SIP Port: | Call Type: | ||
| VoIP State: | Idle | PSTN State: | Idle |
| VoIP Tone: | PSTN Tone: | ||
| VoIP Peer Name: | PSTN Peer Name: | ||
| VoIP Peer Number: | PSTN Peer Number: | ||
| VoIP Call Encoder: | VoIP Call Decoder: | ||
| VoIP Call FAX: | VoIP Call Remote Hold: | ||
| VoIP Call Duration: | VoIP Call Packets Sent: | ||
| VoIP Call Packets Recv: | VoIP Call Bytes Sent: | ||
| VoIP Call Bytes Recv: | VoIP Call Decode Latency: | ||
| VoIP Call Jitter: | VoIP Call Round Trip Delay: | ||
| VoIP Call Packets Lost: | VoIP Call Packet Error: | ||
| VoIP Call Mapped RTP Port: | |||