FreeSwitch - call through registered extension

Sometimes with SIP PBX you have a trunk, that registers on PBX. Most common example - various gateways.
Asterisk has a simple solution for this (as in Asterisk difference between trunk and extension is more cosmetic, than functional), but FreeSWITCH uses concept of gateways.

But it's a very powerful system. So we will make bridge statement by ourselves.

Adoption for FusionPBX.

condition - destination_number - ^(\d{10,20})$
action - set - reged_ext=username_of_extension
action - set - num_to_dial=$1
action - bridge - {absolute_codec_string='${outbound_codec_prefs}'}${regex(${sofia_contact(${reged_ext}@${domain_name})}|(^\w+/\w+)/|%1)}/sip:${num_to_dial}@${regex(${sofia_contact(${reged_ext}@${domain_name})}|(\d+.\d+.\d+.\d+:\d+.*)|%1)}


Original source (rus)

FreeSwitch variable length

Googling does not helps, so, made own bicycle.
Quick inliner to get variable length in FreeSwitch.

In $1 assuming our variable.

dest_len=${lua(~stream:write(tostring("$1"):len()))}

OpenSIPS proxy with auth on external trunk

A new task has arrived.
Make OpenSIPS (actually, here can be Kamailio as well) handle registration to external service, but also provide capability to dial out from other trunks, that are not aware of any credentials of other trunk and know about only OpenSIPS as a trunk. So, in some sort it's b2b.

1. A -> INVITE -> OpenSIPS                     B

2. A              OpenSIPS ->    INVITE     -> B

3. A              OpenSIPS <-    401(7)     <- B

4. A              OpenSIPS -> INVITE (auth) -> B

5. A              OpenSIPS <-      200      <- B

6. A  <- 200 <-   OpenSIPS

Very simplified scheme here. But the main issue here - on INVITE with auth headers you have to increase CSeq number by 1. Actually creating new dialog with side B. But you still have to hold old dialog with side A. And you have dynamically change on requests CSeq number. 
On question, "is there any OpenSIPS-wise way of making that", got an answer 

"Right now OpenSIPs does not support increasing the cseq during UAC authentication. At the end this is a limitation of the a proxy versus a B2B :)"

So, nothing is left, only to change it by hand.

So, parts of the script would looks like

...

#------------ uac related parts

loadmodule "uac_auth.so"

modparam("uac_auth","auth_realm_avp","$avp(uac_realm)")

modparam("uac_auth","auth_username_avp","$avp(uac_username)")

modparam("uac_auth","auth_password_avp","$avp(uac_password)")

loadmodule "uac.so"

loadmodule "uac_registrant.so"

modparam("uac_registrant", "db_url", "mysql://astercc:astercc@localhost/opensips")

modparam("uac_registrant", "timer_interval", 120)

modparam("uac_registrant", "hash_size", 2)

# In DB we're storing info on external services where to register

modparam("uac_registrant", "db_url", DBURL)

...

route {

...

# Handle sequential requests part

if (has_totag()) {

  if (loose_route()) {

    ...

  } else {

    ...

    if (is_method("ACK") && isflagset(AUTH_DONE)) {

      # Process ACK's

      if ($cs == $avp(original_cseq)) {

        route(INCREASE_CSEQ);

      }

    }

    ...

  }

}

...

# CANCEL processing

if (is_method("CANCEL")) {

  # Addition of process CSeq in a case of auth call

  if (isflagset(AUTH_DONE)) {

    # Process CANCEL's to both sides

    if ($cs == $avp(original_cseq)) {

        route(INCREASE_CSEQ);

    } else {

        route(RESTORE_CSEQ);

    }

  }

  ...

}

...

# INVITE processing

if (is_method("INVITE")) {

  ...

  # Here to find out call to external auth trunk

  if (...) {

    $avp(original_cseq) = $cs;

    setflag(IS_OUTBOUND_CALL);

    # Also point, here, if we already know that trunk with username and pass, provider may require specific Form and To fields

    #if (...) {

    #  uac_replace_to("sip:$tU@$rd");

    #  uac_replace_from("sip:$avp(uac_username)@$rd");

    #}

  }

  ...

}

onreply_route[1] {

...

  # On reply just restore original CSeq.

  route(RESTORE_CSEQ);

...

}

failure_route[1] {

  ...

  # Authentication reply on outbound call received?

  if (t_check_status("40[17]") && isflagset(IS_OUTBOUND_CALL)) {

    # Have we already tried to authenticate?

    if (isflagset(AUTH_DONE)) {

      t_reply("503","Authentication failed");

      exit;

    }

    # Set flag of auth preformed

    setflag(AUTH_DONE);

    # Getting realm from responce

    # Get Proxy-Auth header

    if ($(<reply>hdr(Proxy-Authenticate))) {

      $var(raw_auth) = $(<reply>hdr(Proxy-Authenticate));

    } 

    # Prefer WWW-Authenticate to Proxy-Authenticate

    if ($(<reply>hdr(WWW-Authenticate))) {

      $var(raw_auth) = $(<reply>hdr(WWW-Authenticate));

    }

    $var(reg_start) = "/(.*?)realm=\"//g";

    $var(reg_end) = "/\"(.*)//g";

    $var(raw_auth) = $(var(raw_auth){re.subst,$var(reg_start)});

    $avp(uac_realm) = $(var(raw_auth){re.subst,$var(reg_end)});

    # --- Here we assume, that avp(uac_username) and avp(uac_password) are set elsewhere

    if (uac_auth()) {

      route(INCREASE_CSEQ);

    } else {

      exit;

    }

    route(RELAY);

  }

  ...

}

route[RESTORE_CSEQ] {

  if (isflagset(AUTH_DONE) && is_avp_set("$avp(original_cseq)")) {

    remove_hf("CSeq:");

    append_hf("CSeq: $avp(original_cseq) $rm\r\n", "Call-ID"); 

  }

}

route[INCREASE_CSEQ] {

  if (isflagset(AUTH_DONE) && is_avp_set("$avp(original_cseq)")) {

    $var(inc_cseq) = $(avp(original_cseq){s.int}) + 1;

    remove_hf("CSeq:");

    append_hf("CSeq: $var(inc_cseq) $rm\r\n", "Call-ID");   

  }

}

Freeswitch HA mode based on Keepalived.

Idea to have 2 instances of Freeswitch in Master-Slave mode, if one node dies, second will automatically takeover calls with 2-3 seconds lag. Yep, media would be restored.

Main scheme is looks like this

We need 3 IP's. 2 for nodes and 1 for floating.

Solution based on this, but with a bit of additional config.
Used Debian 8x64 as a host OS.
Assume, that we have FreeSWITCH'es already installed.

Install keepalived on both nodes.

apt-get install keepalived

Create files on both nodes

/etc/keepalived/keepalived.conf

global_defs {
    router_id FREESW
}

vrrp_script chk_fs {
    script "/etc/keepalived/scripts/ka-status.pl"
    interval 1
}

vrrp_instance VI_FREESW {
    # replace with SLAVE on a slave node
    state MASTER
    # change to SLAVE on slave node
    interface <YOUR_INTERFACE_HERE>
    #interface eth0
    virtual_router_id <YOUR_ROUTER_ID>
    # virtual_router_id 15
    # higher is preferred for master
    # disable to have failover be sticky
    priority 1
    advert_int 1
    unicast_src_ip <CURRENT_NODE_IP>
    #unicast_src_ip 192.168.10.10
    unicast_peer {
        <SLAVE_NODE_IP>
        #192.168.10.11
    }
    authentication {
        auth_type PASS
        auth_pass YourPassHere
    }
    notify "/etc/keepalived/scripts/ka-notify.pl"
    virtual_ipaddress {
        <FLOATING_IP/CIDR> dev <YOUR_INTERFACE_HERE>
       # 192.168.0.15/24 dev eth0 - Example
    }
    track_script {
        chk_fs
    }
}

/etc/keepalived/scripts/ka-notify.pl
#!/usr/bin/perl

# INSTANCE|VI_FREESW|BACKUP|50
my ($what,$id,$state,$prio) = @ARGV;
open(STDOUT, "|/usr/bin/logger -t ka-notify");

print "what($what) id($id) state($state) prio($prio)\n";

if ( $state eq "MASTER" )
{
    print "Instance went to master, issuing sofia recover.\n";
    system("/usr/bin/fs_cli", "-x", "sofia recover");
}


/etc/keepalived/scripts/ka-status.pl
#!/usr/bin/perl

use Sys::Syslog;
openlog "ka-status", "ndelay,pid", "local0";

my @required = ("internal", "external");

my %saw = ();
open(my $in, "-|") || exec("/usr/bin/fs_cli", "-x", "sofia xmlstatus");
while ( defined(my $line = <$in>) )
{
    if ( $line =~ m|<name>(.*)</name>|o )
    {
        $saw{$1} = 1;
    }
}
close($in);

foreach my $profile ( @required )
{
    if ( ! $saw{$profile} )
    {
        syslog(LOG_INFO, "sip profile $profile not found, marking failure");
        exit(1);
    }
}
exit(0);


chmod +x /etc/keepalived/scripts/*.pl
echo "net.ipv4.ip_nonlocal_bind = 1" >>  /etc/sysctl.conf 

Make sure, that FreeSWITCH'es uses same runtime database and uses same hostname in switch.conf Also profiles you control, need to listen on FLOATING_IP address.
Better reboot here.

service keepalived start

To check state, best is to use ip addr show. On which node FLOATING_IP address is, this is master :)

You can play with priority parameter in keepalived.conf file to make one node Master at all cases.
Point, there is a issue, when you restore calls on one node, you can't get same calls back with sofia recover.

FusionPBX/FreeSWITCH save CLID on transfer

Idea is when call is received, transfer to next destination should come with callerID was received originally, not updated in moment of transfer. Sometimes it's needed for correct CRM integration or peoples just get used to it, cause it's default with blind transfer.

Actually, same ideas that described here, but for use in FusionPBX.

1. Create context save_transfer with order ~85
condition - <empty>
action - export - nolocal:execute_on_answer_1=lua number_save_on_transfer_store.lua
action - export - api_hangup_hook=lua number_save_on_transfer_db_cleanup.lua

2. Create context restore_transfer with order higher, than save_trasfer, ~80
condition - ${db(exists/number_transfer_store/${sip_from_user})} - ^true$
action - set - restored_number_on_transfer=${db(select/number_transfer_store/${sip_from_user})} - inline
action - set - effective_caller_id_number=${restored_number_on_transfer}
action - set - effective_caller_id_name=${restored_number_on_transfer}
action - db - delete/number_transfer_store/${sip_from_user}
condition - ${db(exists/number_transfer_store/${sip_from_user}_name)} - ^true$
action - set - restored_number_on_transfer_name=${db(select/number_transfer_store/${sip_from_user}_name)} - inline
action - set  - effective_caller_id_name=${restored_number_on_transfer_name}
action - db - delete/number_transfer_store/${sip_from_user}_name

Lua files:
/usr/share/freeswitch/scripts/number_save_on_transfer_store.lua

-- Save number_answered / original caller_id to database

--api = freeswitch.API()

if (session:ready()) then
    answered_extension = session:getVariable("dialed_user")

    caller_id = session:getVariable("restored_number_on_transfer")
    caller_name = session:getVariable("restored_number_on_transfer_name")

    if (caller_id == nil) then
        caller_id = session:getVariable("sip_from_user")
    end
    if (caller_name == nil) then
        caller_name = session:getVariable("sip_from_display")
    end
    if (answered_extension ~= nil and caller_id ~= nil) then
        freeswitch.consoleLog("INFO", "[NUMBER_ON_TRANSFER_SAVE] Got answered call from "..caller_id.." to "..answered_extension.."\n")
        session:execute('db', 'insert/number_transfer_store/'..answered_extension..'/'..caller_id)
        if (caller_name ~= nil) then
            session:execute('db', 'insert/number_transfer_store/'..answered_extension..'_name/'..caller_name)
        end
    end
end

/usr/share/freeswitch/scripts/number_save_on_transfer_db_cleanup.lua

-- Cleanup database

api = freeswitch.API()
sip_to_user = env:getHeader("variable_last_sent_callee_id_number")

if (sip_to_user ~= nil) then
--serialized = env:serialize()
--freeswitch.consoleLog("INFO","[hangup]\n" .. serialized .. "\n")

    freeswitch.consoleLog("INFO", "[DB_CLEANUP] Cleaning " .. sip_to_user .. "\n")
    api:executeString('db delete/number_transfer_store/'..sip_to_user)
    api:executeString('db delete/number_transfer_store/'..sip_to_user..'_name')
end



P.S.: At the end, db is can be easily replaced with hash. As we don't need persistence storage here.

Freeswitch/Fusion stop/resume record on the fly

Most common solution to pause/resume recording on Freeswitch is use stop_record_session and after - record_session along with RECORD_APPEND=true.
What is bad with this solution.
1. Only WAV is supported
2. Seems it's broken for now (1.6.18)
Other option to use is
uuid_record mask/unmask

For FusionPBX it will looks like
action - bind_digit_action - local,*1,api:uuid_record,${uuid} mask ${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext},${bind_target}

action - bind_digit_action - local,*2,api:uuid_record,${uuid} unmask ${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext},${bind_target}

So, you may press *1 to stop recording and *2 to resume it. Bad in this solution - you will have pauses in conversation while record is masked.

Freeswitch Postgres database scheme

Freeswitch has a great option to put internal database from SQL to Postgres. But in a case if Postgres DB is BDR (as I've described here), there is need some additional work on this database.

So, I've created SQL file to use, based on DigiDaz file. 

You can find it here. Don't forget to set

<param name="auto-create-schemas" value="false"/>

<param name="auto-clear-sql" value="false"/>

in your switch.comf.xml file.

Also few things about BDR. You can't add primary key to existing database in one command. But you can use thing like:

ALTER TABLE tbl ADD COLUMN tbl_uuid uuid;

ALTER TABLE tbl ALTER COLUMN tbl_uuid  SET DEFAULT  gen_random_uuid();

ALTER TABLE tbl ADD PRIMARY KEY (tbl_uuid);