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spa 3000 : how to send the PSTN call to both Line 1 and a SIP uriTechnical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
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| I am using a SPA 3000. PSTN line is connected to FXO port and an analog phone is connected to FXS port. I want the phone on FXS port ring whenever PSTN line receives a call from outside. I also want to make a SIP uri (softphone on my PC) to receive the same call (and thus ring) is this setup possible ? thanks... |
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So, to answer the question, on an incoming PSTN call you do can do one after the other, not simultaneously (not at the same time). If you want to ring both phones at the same time you could setup the SPA3000 to send the call to a PBX system like pbxes.com which via ring groups and extensions could do what you suggest. Calling a sip uri instead of a regular number can present some complications, but it can be done. |
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| thank you for the answer. Now I forward the call to a SIP uri via the PSTN-to-Voip DP. However, I want to ring the phone on Line 1 if that SIP uri is not reachable and/or if it does not answer. How can I do that ? |
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| Surprisingly enough, I was thinking about this as well in preparation for my proposed installation of an Asterisk box. Correct me if I'm wrong: For each action that occurs on a line, a signal in the form of a tone is sent back to the SPA, e.g busy, disconnect, etc. Now, if we could have a program that used, say, loops, that would loop through a list of pre-defined gateways, trying each one after the other if the previous one returned a busy/disconnect tone, would that not do the job? So, I was thinking, the SPA can't do this itself, so maybe an external device? I think commercial PBX's do this, I know many offices have the system try a series of numbers until a reply (or if not then voicemail) is reached. Asterisk, from what I've heard is quite powerful, more than what it may seem like on the surface. Also, I'm not sure but can you get plugins for it? Also, as it's 'officially' run on Linux, and Linux is traditionally a programmer's heaven, couldn't you write a program in C that could interface with the SPA and detect the signals returned to it, and then do the looping using an Asterisk gateway/group list? Obviously, the SPA would need to interface with Asterisk, and be able to send/receive signals/commands from it, but surely this bit is possible? Please note, my C programming skills are not the best in the world, and I'm trying to think about this in a modular manner and 'out of the box'. I'm really getting into the whole idea of VoIP, and if anyone wanted to work more on this, or investigate such user-written programs such as those discussed here, I'd be up for the challenge. Hope that's food for thought. Hussein. |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| Point to SIP URI for SPA3000 phone rang but can't hear either side | hok | IPKall Support Forum | 3 | October 18th, 2006 04:35 PM |
| Receive Sip Uri Calls + Call Forwarding | cunyalen | Linksys (Sipura) VoIP Support Forum | 0 | October 3rd, 2006 04:13 PM |
| SPA3102 forward sip call to PSTN line | hafeezc | Linksys (Sipura) VoIP Support Forum | 16 | August 19th, 2006 02:14 PM |
| 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 |
| can't switch a call on line 1 to pstn line on spa 3000 | palex9 | Linksys (Sipura) VoIP Support Forum | 3 | May 8th, 2005 01:46 AM |