05-28-2013 10:03 AM - edited 03-21-2019 10:03 AM
I an trying to get two SPA's setup so one side is receiving a constant audio (paging audio from a phone system) and the other is connected to a page amp to deliver the audio. I have been told and have found blog posts that this is possible, but I cannot find the information to make it work. I have the master setup as a SAS (with auto answer on) and the slave setup to hot dial the master, but the reply to the invite is 503: Service Unavailable. Does anyone happen to have this working and how do I set it up? I only need one way audio, so this should be fairly simple. Thanks! - Jeremy
05-29-2013 07:27 AM
Figured it out myself, I forgot to use a transformer to keep the port off hook. But I did discover you have to have the rtp sink setup or the other side will beep every 10 seconds or so. - Jeremy
10-18-2013 05:45 PM
Hi Jeremy,
I have a similar problem, I have one PSTN line (say Line1) with free minutes to mobiles, so its good for outgoing calls. The other line (say Line2) which i have is acually VoIP but it comes with its own hardware (magicJack if you have heard) so I can't use a SIP client and have to use the supplied Hw client, but it does give me an option to connect any normal phone to this magicJack (i suppose that would make it a fxs port). Now this magicJack is cheap for other people to call me.
I want to find a solution so that all the calls I receive on Line2 get forwarded to my mobile number via Line1. And if I receive any calls on Line1 they should be treated normally (my home phone rings). Do you have some idea how I can achieve this with minimal spend? Thanx
Atif
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