<div dir="ltr"><div><div>Hi <span name="Shaw Xu" class="">Shaw Xu,<br><br></span></div><span name="Shaw Xu" class="">Provide some asterisk and freeswitch logs for debugging ? .<br><br></span></div><span name="Shaw Xu" class="">Regards<br>
<br>Varghese Paul <br></span></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jul 17, 2014 at 6:58 AM, Nandy Dagondon <span dir="ltr"><<a href="mailto:nandy1925@gmail.com" target="_blank">nandy1925@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div class="gmail_extra">I used to work with */Elastix before ... so, I reset as an Asterisk newbie. Here's my input.</div>
<div class="gmail_extra"><br></div><div class="gmail_extra">As long as FS and Asterisk register to each other, FS is versatile enough to send any SIP URI to Asterisk and vice-versa. It's up to */Elastix Inbound extensions (dialplan) to route them to your clients. You can expand the Wiki simple extension-to-extension setup to suit your needs. Specific problems? Just lay them out here. We'll take a look.</div>
<span class="HOEnZb"><font color="#888888">
<div class="gmail_extra"><br></div></font></span><div class="gmail_extra"><span class="HOEnZb"><font color="#888888">
/Nandy<br><br></font></span><div class="gmail_quote"><div><div class="h5">On Wed, Jul 16, 2014 at 7:33 AM, Shaw Xu <span dir="ltr"><<a href="mailto:shawx8@gmail.com" target="_blank">shawx8@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">
<div dir="ltr"><div class="gmail_extra"><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">I am newbee of freeswitch.( many year experience on asterisk / Elastix / freepbx) </div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">
<br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">I checked freeswitch is great as a soft switch(means works as the VoIP router for many PBXes) , and Asterisk (Elastix / freepbx / PIAF and so on) have more feature as the pbx.<br>
</div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">So my idea is using freeswitch as the the main hub ( to SIP trunk, which will provide hundreds DID to many PBXes like asterisk / freepbx/elastix.) and freeswitch will collect all minutes to IMS (sip trunk provider) </div>
<div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">I checked forum, looks the wiki only has documents on FreeSwitch and Asterisk directly connect(dial exten each other) , which my intention is using Elastix / PIAF (on top of freepbx) as sub PBX, and the call routered from FreeSwitch should go through the Elastix inbound Routes rules instead the direct extension as in the wiki <a href="https://confluence.freeswitch.org/display/FREESWITCH/Asterisk" style="font-family:arial;font-size:small" target="_blank">https://confluence.freeswitch.org/display/FREESWITCH/Asterisk</a></div>
<div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">system environment is CentOS 5.10 with follow the wiki install guide to install freeswitch.( by the way, the MOH not working after install the MOH sound, because the directory is not correct, I copy to the right location manually, and need start the moh for first time.) </div>
<div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">Any ideas are appreciated.</div><br><br></div></div>
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