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<p class="MsoNormal"><span lang="EN-US">Hi,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I need a help…<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I need to build a small system with amateur simplex links with low cost and low power nodes with any PBX functions (gsm trunk, voice menu, mailbox). PBX server based on XIPPR, but may be a Freeswitch too. I need to
use only SIP protocol. System programmed using inbond DTMF.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Hardware may be a Raspberry pi, with USB soundcard for audio and USB-UART adapter for radio control.
<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Idea is a DTMF control. Any Asterisk app or module need to code and decode dtmf digit (letter)<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">PTT activate with DTMF “A” and deactivate with DTMF “B”. For example, node number 2000 connected to node 3000. If to node 3000 audio input comes DTMF letter A (from radio channel), it will activate PTT on node 2000 and
letter B will deactivate PTT. <o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">One UART pin can be used as PTT, second as CarrierDetect. If CD pin grounded (radio channel buzy) node generate DTMF tone A for activating PTT of second side radio (or radios, if used conference). If CD pin ungrounded,
node generate a B tone. <o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">This system will be very simple, possible to use different server platforms, gateways, node cost is Raspi 40eur, USB audio and UART is 8eur<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Same system in Freeswitch tested and works very fine!<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Could any help with making this addition module? I think, it may be interesting for other voip/roip users<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Thanks,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Mihhail<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">ES1BIS<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
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