[Freeswitch-users] CISCO FXO - IVR

Oliver Schenk olimonkey at gmail.com
Tue Oct 11 18:23:01 MSD 2011


Hi All,


I have posted here some months back regarding interfacing CISCO with
FreeSWITCH. I have had basic success, however there is quite a lot of
fine tuning to do which is beyond my general knowledge or what I can
find via Google.


Currently I can do the following:


1. Make a call from any phone to a specific phone number. This phone
number ends up at the CISCO FXO card. This then redirects to a
specific extension at the FreeSwitch box. The dialing plan then causes
a managed C# .NET application to start playing.

2. In the other direction, I have a windows service that monitors a
given database. If any alarm conditions occur it uses ESL and the
“Originate” API call to create a connection to the CISCO. The CISCO is
then configured to redirect the given number back through the FXO card
via plain old telephone (POTS).

So, I can successfully make incoming and outgoing connnections on demand.




PROBLEMS:

1. Using ESL for outgoing call, I don’t know how to determine whether
a) the phone on the far end was picked up,   b) whether the phone on
the far end just rang out   c) whether it was busy. In all cases the
IVR just starts playing right away even if I don’t even pick up the
phone. (I THINK THAT HAS SOMETHING TO DO WITH CISCO CONFIG). It seems
that FreeSwitch simply triggers the CISCO to make a call, but then has
no ability to track progress and assumes a connection was made. How do
I get POTS to talk to VOIP/SIP to talk to FreeSwitch?

2. Incoming calls simply come in as “Processing unknown
<0000000000>->1024 in context public. The actual incoming phone number
does not seem to be getting from the CISCO to the FreeSwitch box.

3. I have 2 lines. I was simply going to assign extension 1024 for
line 1 and 1025 for line 2. But for outgoing calls from my managed
code how can I use ESL to say:

a. Is line 1 busy? Then use Line 2 else make call.

b. Is line 2 busy? Then wait for next scan else make call.






My Code in the outgoing service:

ESLconnection eslConection = new ESLconnection(“x.x.x.x”, “8021”, “password”);

If (eslConnection.Connected() == ESL_SUCCESS)
{
    eslConnection.Api(“originate”, “sofia/internal/0” + phoneNumber +
“@” + CISCO_IP + “ ‘&managed(xNetIVR)’”);

    // determine if call is ringing, busy or picked up here
    // no idea how to do that…
}






Relevant CISCO config:


version 12.4
!
!
!
voice service pots
fax rate disable
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
sip
registrar server expires max 3600 min 3600
  no update-callerid
  no call service stop
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
mode cme
source-address <CISCO IP> port 5060
max-dn 10
max-pool 10
authenticate register
hold-alert
tftp-path flash:
!
!
!
voice-card 0
dsp services dspfarm
!
!
!
voice-port 0/3/0
output attenuation -3
no comfort-noise
cptone AU
connection plar opx 1024
impedance complex1
caller-id enable
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
!
!
!
dial-peer voice 10 pots
incoming called-number .T
no digit-strip
direct-inward-dial
port 0/3/0
forward-digits all
!
dial-peer voice 100 pots
preference 1
destination-pattern 0T
no digit-strip
port 0/3/0
forward-digits all
!
dial-peer voice 1 voip
destination-pattern 1024
session protocol sipv2
session target ipv4:<FreeSwitch IP>
session transport tcp
dtmf-relay rtp-nte
codec g711alaw
no vad
!
!
sip-ua
presence enable
!
!




Relevant FreeSwitch config:


conf\dialplan\public\00_xNetIVR.xml and conf\dialplan\default\00_xNetIVR.xml:

<include>
  <extension name="testapp">
  <condition field="destination_number" expression="^(1024)">
     <action application="managed" data="xNetIVR"/>
  </condition>
</extension>
</include>





conf\directory\default\1024.xml

<include>
  <user id="1024">
    <params>
      <param name="password" value="xxxx"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1024"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1024"/>
      <variable name="effective_caller_id_number" value="1024"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
    </variables>
  </user>
</include>




Public.xml and default.xml dial plans are basically the same as
default installation.







Cheers,



Oliver Schenk



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