3rd Party SIP client and Session Manager (bypass CM)

[hanbalfrek]

Hi all,

I hear this is possible, but am having problems getting it to work.

Has anyone registered a 3rd party soft phone to the session manager without either the Avaya CM or Nortel CS1K as a feature/evolution server?

From a feature perspective, I just want to be able to make and take calls.

Any help would be appreciated.

 

[joesena]

I had set up FreeSwitch and connected from it to CM (no Avaya Session Manager) via a SIP trunk. I registered a SIP phone to FreeSwitch and programmed CM and FreeSwitch to send calls to and from that phone. It worked well. Let me know if you need specifics.

 

[damnplanetearth]

Hi Joesena,

can you please post some screenshots/settings of CM & FS config, thanks in advance.

 

[pabxdude]

i do this all the time when testing SBC's.

i use X-Lite.

simply add a user in SMGR. the main thing you DO NOT want to do is add an Originating and terminating application sequence to the station.

You still add a SIP handle.

Then in X-lite just add the username and password (which is the communication profile password) and enter the registration address as the SM100

calls in and out work fine as long as your SMGR dial plans exist.

 

[damnplanetearth]

Hi Mate,
thanks for your help, would this let me do multiple calls as iam planning to have around 20 to 30 simultaneous calls. so I was thinking to link directly to CM rather than using SM. if SM is last option then I would setup a trunk rather than station, would that be best way?
thanks

 

[joesena]

Below is the freeswitch.xml from the setup I had for this. (I feel like I had a gateway called "phones" and the directory entries pointed to that, but it's not in here, and I can't test right now). I *think* you should be able to install a clean install of FreeSwitch and just replace the freeswitch.xml in the conf folder with this one (I did a lot of ripping out of mods on mine as well, but you shouldn't need that)

There are a few places I've specified where you'll need to enter the IP address of your CM, and down the bottom, I've made a space or two where you would enter the extensions of the phones which would register to FreeSwitch.

Additional work is required on the Avaya side. You'll have to set up UDP and AAR to get calls over to the SIP Trunk to the FreeSwitch. You'll also need to set up the signalling group and trunk to the FreeSwitch. Finally, you'll need to set up the station on the Avaya side. You can either set up a new station of type 9620SIP and put the trunk on page 6, or set up off-PBX station mapping on an existing station (application OPS).


<?xml version="1.0"?>
<document type="freeswitch/xml">
<section name="configuration" description="Various Configurations">

<configuration name="switch.conf" description="Core Configuration">
<settings>
<param name="colorize-console" value="true"/>
<param name="max-db-handles" value="50"/>
<param name="db-handle-timeout" value="10"/>
<param name="max-sessions" value="1000"/>
<param name="sessions-per-second" value="30"/>
<param name="loglevel" value="debug"/>
</settings>
</configuration>

<configuration name="console.conf" description="Console Logger">
<mappings>
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
</mappings>
<settings>
<param name="colorize" value="true"/>
<param name="loglevel" value="info"/>
</settings>
</configuration>

<configuration name="sofia.conf" description="Sofia Endpoint">
<profiles>
<profile name="p_default">
<domains>
<domain name="all" alias="false" parse="true"/>
</domains>
<gateways>
<gateway name="avaya">
<param name="realm" value="xxx.xxx.xxx.xxx"/> //***REPLACE with your Avaya's IP address e.g. 192.168.1.1
<param name="register" value="false"/>
<param name="register-transport" value="tcp"/>
<param name="contact-params" value="tport=tcp"/>
</gateway>
</gateways>
<settings>
<param name="sip-port" value="5060"/>
<param name="rfc2833-pt" value="101"/>
<param name="dialplan" value="XML"/>
<param name="context" value="d_default"/>
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="codec-prefs" value="PCMU"/>
<param name="disable-transcoding" value="true"/>
<param name="inbound-late-negotiation" value="false"/>
<param name="caller-id-type" value="rpid"/>
<param name="force-register-domain" value="$${local_ip_v4}"/>
<param name="enable-timer" value="false"/>
<param name="minimum-session-expires" value="120"/>
</settings>
</profile>
</profiles>
</configuration>

<configuration name="modules.conf" description="Modules">
<modules>
<load module="mod_console"/>
<load module="mod_sofia"/>
<load module="mod_commands"/>
<load module="mod_dptools"/>
<load module="mod_dialplan_xml"/>
</modules>
</configuration>
</section>

<section name="dialplan" description="Regex/XML Dialplan">
<context name="d_default">
<extension name="in from avaya">
<condition field="network_addr" expression="(xxx\.xxx\.xxx\.xxx)"/> //***REPLACE with your Avaya's IP address e.g. 192.168.1.1 becomes 192\.168\.1\.1 here
<condition field="destination_number" expression="(\d{4})$">
<action application="bridge" data="sofia/p_default/$1%$${local_ip_v4}"/>
</condition>
</extension>
<extension name="out from phones">
<condition field="destination_number" expression="(\*?\d+)">
<action application="bridge" data="sofia/gateway/avaya/$1"/>
</condition>
</extension>
</context>
</section>

<section name="directory" description="User Directory">
<domain name="$${local_ip_v4}">
<variables>
<variable name="default_gateway" value="avaya"/>
</variables>
<user id="xxxx"> //***REPLACE with the extension of your phone e.g. 1000
<params>
<param name="password" value="whateverpasswordyouwantforthephone"/>
</params>
</user>
<user id="xxxx"> //***REPLACE with the extension of the next phone e.g. 1001
<params>
<param name="password" value="whateverpasswordyouwantforthephone"/>
</params>
</user>
</domain>
</section>
</document>

 

[damnplanetearth]

thanks, I will test next week and revert