Pls see my comments below. Hope it helps.


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
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
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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