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