<div dir="ltr">Pls see my comments below. Hope it helps.<br><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div><a href="https://camiguintelekom.wordpress.com/about-the-author/" target="_blank">Nandy Dagondon</a></div><div dir="ltr"><br></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div>
<br><div class="gmail_quote">On Tue, Apr 25, 2017 at 9:23 PM, Valter Nogueira <span dir="ltr"><<a href="mailto:valter@fastway.com.br" target="_blank">valter@fastway.com.br</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"><div>I have an app that runs in Asterisk and I am trying port it to FS.</div><div><br></div><div>Below, I pasted a small sip.conf that sums up all my needs.</div><div><br></div><div>1. We register out in a VOIP_PROVIDER</div><div><br></div><div>2. An external asterisk register in our asterisk as CLIENT01</div><div><br></div><div>3. Calls from CLIENT01 route to VOIP_PROVIDER</div><div><br></div><div>4. Yealink's, with ACL restrictions, also uses VOIP_PROVIDER to external calls</div><div><br></div><div>5. VOIP PROVIDER and Yealinks use one NIC (eth0)</div><div><br></div><div>6. CLIENT01 uses another NIC (eth1)</div><div><br></div><div>In Asterisk all configs are keep together in sip.conf.</div><div><br></div><div>However, in FS it seems I should spread out things in different configs parts.</div><div><br></div><div>So:</div><div><br></div><div>Should Yealink's accounts go into /usr/local/freeswitch/<wbr>conf/directory/default?</div></div></blockquote><div>No need since Yealink is registered to VOIP_PROVIDER, right? Your public dialplan should take care of the incoming calls - e.g. destination_number</div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><br></div><div>Should I put VOIP_PROVIDER in sofia's external profile?</div></div></blockquote><div>Yes. </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><br></div><div>Should I create an additional external profile to CLIENT01? Where should I define CLIENT01 account to allow it register in? In /usr/local/freeswitch/conf/<wbr>directory/default?</div></div></blockquote><div>CLIENT01 in directory/default but ... Yes, create a new profile so that it will listen with the WAN IP address and new port (e.g. WAN_IP:5070 since 5060 listens to LAN clients registering to Freeswitch, port 5080 for FreeSwitch to register to VOIP_PROVIDER. Search for "doublenat" profile example in the Wiki. </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><br></div><div>Thanks for any help.</div><div><br></div><div>Valter</div><div><br></div><div>=== SIP.CONF ===</div><br clear="all"><div><div class="m_6249630823480443833gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div>[general]</div><div>disallow=all</div><div>context=default ; Default context for incoming calls</div><div>allowoverlap=no ; Disable overlap dialing support. (Default is yes)</div><div>bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)</div><div>bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)</div><div>srvlookup=yes ; Enable DNS SRV lookups on outbound calls</div><div>alwaysauthreject=yes</div><div>allowguest=no</div><div><br></div><div>;voip provider</div><div>register => 185999:1010@xxx.xxx.xxx.xxx:<wbr>5060</div><div><br></div><div>[VOIP_PROVIDER]</div><div>type=peer</div><div>context=FDTRONCO_94</div><div>host=xxx.xxx.xxx.xxx</div><div>port=5060</div><div>username=185999</div><div>authuser=185999</div><div>authname=185999</div><div>secret=XXXXXXXXXXXXXX</div><div>fromdomain=xxx.xxx.xxx.xxx</div><div>fromuser=185999</div><div>insecure=port,invite</div><div>canreinvite=no</div><div>nat=no</div><div>disallow=all</div><div>allow=ulaw:30</div><div>dtmfmode=rfc2833</div><div>ignoreregexpire=yes</div><div>language=pt_BR</div><div>call-limit=9999</div><div><br></div><div>;EXTERNAL ASTERISK ACCOUNT - CALLS GOES THRU VOIP_PROVIDER</div><div>[CLIENT01]</div><div>type=peer</div><div>context=FDTRONCO_60</div><div>host=dynamic</div><div>port=5060</div><div>username=CLIENT01</div><div>authuser=CLIENT01</div><div>authname=CLIENT01</div><div>secret=XXXXXXX</div><div>insecure=port,invite</div><div>canreinvite=no</div><div>nat=yes</div><div>disallow=all</div><div>allow=g729</div><div>dtmfmode=rfc2833</div><div>ignoreregexpire=yes</div><div>language=pt_BR</div><div>call-limit=440</div><div><br></div><div>;yealinks ip phones</div><div>[1000]</div><div>type=friend</div><div>secret=XXXX</div><div>host=dynamic</div><div>deny=<a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a></div><div>permit=<a href="http://192.168.0.0/16" target="_blank">192.168.0.0/16</a></div><div>username=1000</div><div>context=FASTDIALER_RAMAIS</div><div>callerid=testefone <1000></div><div>requirecalltoken=no</div><div>nat=no</div><div>canreinvite=no</div><div>qualify=yes</div><div>disallow=all</div><div>allow=alaw</div><div>call-limit=2</div><div>dtmfmode=RFC2833</div><div>language=pt_BR</div><div><br></div><div>Callgroup=1</div><div>pickupgroup=1</div><div><br></div><div>[2012]</div><div>type=friend</div><div>secret=XXXX</div><div>host=dynamic</div><div>deny=<a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a></div><div>permit=<a href="http://192.168.0.0/255.255.0.0" target="_blank">192.168.0.0/255.255.0.0</a></div><div>username=2012</div><div>context=FASTDIALER_RAMAIS</div><div>callerid=Telefone sem fio <2012></div><div>requirecalltoken=no</div><div>nat=no</div><div>canreinvite=no</div><div>qualify=no</div><div>disallow=all</div><div>allow=alaw</div><div>call-limit=2</div><div>dtmfmode=RFC2833</div><div>language=pt_BR</div><div><br></div><div>Callgroup=1</div><div>pickupgroup=1</div><div><br></div><div><br></div></div></div></div></div></div></div></div></div></div></div>
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