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PSTN to voip gateway..access based on CLITechnical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
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| Hi I am hoping you can help with a feature that I am trying to enable. The spa3102 unit is installed at home and is working fine on VoIP and PSTN for both I/C and O/G, however when I am out with my mobile I would like to dial in .. receive secondary dial tone, then dial out over the VoIP to obtain cheaper long distance calls while on the move. Crucially its only my mobile I want this to happen on, all other callers must just ring the house phone as normal. Now while I can enable this by setting: pstn answer delay to say 2, caller auth method to none, pstn access list to my mobile, .........unfortunatley all callers get secondary dial tone and can dial out over VoIP. I have tried setting the the pstn answer delay to 30 but this just changes the answer timeout for BOTH the mobile and other callers. I have tried numerous setting within the PSTN line config page but cannot get the 3102 to treat my mobile and all other callers differrently with regard to giving access onward dialling to VoIP. Any help would be appreciated regards Chris I have the following: Product Information Product Name: SPA-3102 Serial Number: FM600G504062 Software Version: 3.3.6(GW) Hardware Version: 1.4.5(a) MAC Address: 000E08CD861B Client Certificate: Installed Customization: Open |
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| Hi Thx for the info, I tried that but no joy, have u got this working yourself .. or does anyone else on this forum use it, it would be good to here that is actually a working feature. I reset the the unit to get rid of any rogue config.. set the basic line parameters for sipgate... set dial plan in line settings set the cli in pstn caller ID pattern but still it doesn't work as previously mentioned I used pstn access list but the only time I get the unit to aswer the call is when the "pstn ans delay" is reduced to say 5 but once again this is for all calls not just my mobile. any further help or confirmation that this feature does work would be good thanks Chris |
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| Hi, I have the following setup and it works for me: Mine has PSTN Answer Delay = 16 PSTN Caller ID Pattern is my mobile number. Make sure that your mobile number is OK. In order to debug it try making a phone call with your mobile to the device and look at the "Last PSTN Caller" in the Info Tab under the "PSTN Line Status"on the device. After confirming you should insert this EXACT number in the "PSTN Caller ID Patern" field. Let me know how it works for you. |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| how to configure userid based on caller id for pstn to voip calls? | xbipin | Linksys (Sipura) VoIP Support Forum | 0 | January 21st, 2008 11:57 AM |
| [spa2102-na] can't access web-based utility | dfwdraco76 | Linksys (Sipura) VoIP Support Forum | 4 | September 8th, 2007 01:32 PM |
| sipura 3000 PSTN to VOIP (based on Caller ID) | tisom | Linksys (Sipura) VoIP Support Forum | 16 | March 18th, 2006 01:59 AM |
| PSTN Gateway tone and CLI workarounds required | nish | Linksys (Sipura) VoIP Support Forum | 0 | October 6th, 2005 02:01 PM |
| PSTN-TO-VOIP & VOIP-TO-PSTN Gateway issues | redmat | Linksys (Sipura) VoIP Support Forum | 9 | September 13th, 2005 09:14 PM |