I followed many how to on internet, and nothing work, I requested help from many companies (I use the Synology Asterisk Package witch is probably not working efficiently as this is the 6 version only (Asterisk 1.8.13, Asterisk GUI 2.0.3) : devices do not control what they do, software do not check parameters and do not improve efficiency by automatically correcting errors… Humans are not known to make improvements, they can just test different things and compare, and they are source of errors and jealousy…
I am developping a new voice server for EJFJ Corporation, inventor of worlwide standardizations (Medical Pack, Time Pad Scheduler, Staff COM International, …). I need your help to solve problems, Asterisk version 2.0.0 need to be accessible very soonly, nowaday solutions are totally unacceptable, unefficient, too unsecure, requesting too much knowledge from the administrators, and good things cannot be realized (the best things are fully unreachable).
I have now a PSTN line +41215667478 connected to the Linksys SPA-3102, the Asterisk Server cannot be accessed by this line. I defined it as a proxy in the SPA=>Voice=>PSTN Line, with the identifier, and the password.
SIP Trunk : Rejected PSTN sip 1-pstn 10.0.0.6,dynamic
I deleted the dynamic, as all device have fixed IP address, given by the modem/router, and the SPA-3102 is used as a bridge.
I have a VOIP subscription, +41325133791 connected by Asterisk, the status seems ok (Registered) :
SIP Trunk : Registered SIPCall sip 41325133791 free2.voipgateway.org
I configured everything, more than everything, but details can block, one single parameter can totally block the whole Asterisk system, this is a house of cards, not a strong castle! This is a game for administrators who earn their life badly while making this work, this is totally not satisfying for companies… The Asterisk second version need to be self efficient, the administrator will just ned to explain Askerisk how he wants to use the connected/discovered devices, and how he want to cumulate the specific functionnalities of each part of the Asterisk Systems, to fully satisfy the needs of the users… The language recognition need for example to be vocally recognized whenever the caller speaks, the system use the same language to reply, and give him informations about what Asterisk is doing to process the call efficiently, without a key press menu for language selection (lost of time, unefficient, uneffective, unequitable, illegal)!
When I call the fix number (PSTN, the connected phone ring normally, and I can get the call… When I call the SIP number +41325133791, I hear a long biip and nothing more, the call is closed… Beside that, there are only problems, but I will go step by step… so for now, Asterisk is 2.4% efficient… It requested 3 weeks, to arrive to this point… Thank you for your help, all talents are requisitionned by the international justice, in the EJFJ Penal Affairs… Thank you for those who help us go forward… After, only Digium will benefit, as all lacks and defficiencies will need to be corrected for the 2nd version witch should exist since 1994! Yours faithfully…