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PSTN line Configuration on SPA3102Technical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
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Post your configuration saving it right after your call fails. With your web browser login to the adapter under admin/advanced and save the web page to your hard drive using Save Page As with your web browser. Your passwords are not saved. Then make a .zip file of the saved page. If you are using WinXP you can make a zip file by using Windows Explorer, selecting the saved file and using File ... SendTo ... Compressed (Zip) Folder. Then attach the zip folder to a posting. |
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| I'm not sure I understand the problem. Your configuration looks OK for normal conversation with your VoIP provider. The adapter is aware of it's external ip address. The INFO tab shows the last call dialed and the last call received. The INFO tab shows RTP packets both sent and received for your calls. You didn't get any sound? The RTP packets contain the sound. You have a pstn line attached to the "Line" jack however you don't appear to have any way to use the pstn line. You don't have "Ring Thru To Line 1" enabled or the pstn-to-VoIP gateway enabled. You don't have anything in your dial plan to access the pstn line with a dialed number. I'm not sure it is a problem, but on the Sip Tab you have Handle VIA received: YES, and Handle VIA rport: YES. I usually have them set to NO. |
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| Do outgoing calls work correctly? If you look at the INFO tab on the SPA3102 before and after an incoming call where you hear nothing does the number of RTP Packets Sent: and the number of RTP Packets Recv: increase? In other words does the adapter both send and receive RTP packets during the call? |
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When your adapter responds to an incoming call request (sip invite) the sip invite response tells the calling server the address and port number where it wants to receive the incoming rtp packet stream. I made a previous suggestion that I would try. On the Sip Tab you have Handle VIA received: YES, and Handle VIA rport: YES. Set them both to NO and try the test. If that doesn't help then I would try port forwarding the rtp port range in your router to your adapter. You can find the rtp port range on the sip tab. |
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When your adapter responds to an incoming call request (sip invite) the sip invite response tells the calling server the address and port number where it wants to receive the incoming rtp packet stream. I made a previous suggestion on the Sip Tab you have Handle VIA received: YES, and Handle VIA rport: YES. Set them both to NO and try the test. If that doesn't help then I would try port forwarding the rtp port range in your router to your adapter. You can find the rtp port range on the sip tab. |
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| Okay, I have done the changes to the SPA3102 and also tried the port forwarding, but still getting the same results. I also have a PAP2 box for Vonage conected to the same router, and that works both ways with no problem. Don't these devices work similar? Could my SPA3102 be faulty? |
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| Ha Ha!! I figured it out! The problem was that I forgot that this box is connected through a SSL type VPN. It appears that that type of VPN does not work well with RTP, I may be wrong. I tried another type of VPN, ie, L2TP and I now able to make and receive calls. Does anyone know about RTP and SSL VPN to advise if really does not work? |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| SPA3102 Configuration | VOIP to PSTN | kadeshbailey | Linksys (Sipura) VoIP Support Forum | 16 | March 26th, 2008 08:39 AM |
| SPA3102 - PSTN Line impedance | mdvazeer | Linksys (Sipura) VoIP Support Forum | 1 | March 24th, 2008 04:32 PM |
| Spa3102: PSTN Line and incoming call | cuordileone | Linksys (Sipura) VoIP Support Forum | 4 | January 15th, 2008 06:58 PM |
| BLF on phone when SPA3102 PSTN line is in use | Dox | Linksys (Sipura) VoIP Support Forum | 0 | July 27th, 2007 06:41 PM |
| SPA3102 Doesnt see/pick up PSTN line | Calorno | Linksys (Sipura) VoIP Support Forum | 6 | April 18th, 2007 10:54 PM |