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SPA-3K VoIP-to-PSTN: SIP 410 Gone messageTechnical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
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| Hi All, I try to configure my SPA3000 to act as a 1-Port FXO gateway with a pbx only for calls from VoIP to PSTN in France. My pbx sends INVITE to the SPA without authentication (I put my pbx IP address into VoIP Access List). The SPA returns me a 410 Gone message immediately (even with 5 seconds for PSTN Dialing Delay) that makes me suppose there is no off hook… All calls from my pbx (through ‘Voip2’ based on documentation) have to be routed directly to the PSTN. My pbx manages the dial plan. I think I don’t need any dialplan because I don’t use Line1.As I understand from the sipura documentation, it is possible to make calls from VoIP to PSTN without authentication (but not from PSTN to VoIP). Here are some of the parameters I put : PSTN Line Tab : * In SIP Settings section : - SIP Port: 5060 * In VoIP-To-PSTN Gateway Setup section : - VoIP-To-PSTN Gateway Enable: yes - VoIP Caller Auth Method: none - One Stage Dialing: yes - Line 1 VoIP Caller DP: none - VoIP Caller Default DP: none - Line 1 Fallback DP: none - VoIP Caller ID Pattern: - VoIP Access List: <ip address of my PBX which send the INVITE> or *.*.*.* * In PSTN-To-VoIP Gateway Setup section - PSTN-To-VoIP Gateway Enable: no Line 1 Tab : - Line Enable: no Thanks for help, PO |
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| PO: ¿Do you mean your PBX is a software one?, ¿Is it asterisk?. If it is, then there is a Wizard for configuring SPA3000 for asterisk here http://voxilla.com/spa3kasterisk.php . If it is not, please let me know to take another alternative. Juan C. |
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| Hi Juan, it is a software one but not asterisk. the following assumptions: Line 1 will be connected to Asterisk as a normal extension is not possible as my product doesn't support authentication for sip trunking to gateways and I have to use least cost routing on the pbx supporting gateways and not extension to route. cheers, PO |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| Passing a SIP URI to the VoIP line from PSTN SPA 3000 | marsaro | Linksys (Sipura) VoIP Support Forum | 0 | July 7th, 2006 05:50 AM |
| SPA3000 and *, plz help to Dial SIP to PSTN and PSTN to SIP | poi9m | Linksys (Sipura) VoIP Support Forum | 0 | June 29th, 2006 12:37 PM |
| PSTN-> VoIP Gateway with multiple SIP Servers? | mvandersluis | Linksys (Sipura) VoIP Support Forum | 0 | March 29th, 2006 07:12 PM |
| message waiting indication and SIP Subscribes | SDHank | BroadVoice Support Forum | 0 | March 28th, 2005 06:08 AM |
| VoIP to PSTN Calls- VoIP2 sends wrong sip answer | amsco | Linksys (Sipura) VoIP Support Forum | 1 | September 7th, 2004 05:09 PM |