[Freeswitch-users] from Asterisk sip.conf to FS

[Nandy Dagondon]

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 at 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 at 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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