Re: [Freeswitch-users] Sangoma Media Gateway

18 Aug 2011

      Hi Ben

Glad I could help.
I was also very new to FS when I setup SMG and took me quite a while to figure out how SMG is put together, etc...

Thanks to the excellent help and guidance from the Sangoma Support Team I could get my system up and running to my requirements.

Regards
Deon

On Aug 18, 2011, at 8:37 AM, Ben Naylor wrote:
...
Hi Deon
This is great, just what I was looking for!
I am in Germany implementing this over the next few days (or however long it takes), so will start to play around with it today.
Many thanks for your help, very much appreciated!
Regards
Ben
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Deon Vermeulen
Sent: 18 August 2011 07:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sangoma Media Gateway
Hi Ben
I do it as follows.
Go to File Editor -> dialplan -> default.xml
Look for  <extension name="to-sip">
At the very end of the extension there is a part that already shows you how to route to IP.
<!-- uncomment this if you want to dial to a fixed IP addr -->
                                    <!-- <action application="bridge" data="sofia/internal/${destnumber}@192.168.1.1"/> -->
So you could have something like this.
<action application="set" data="sip_contact_user_replacement=${destnumber}"/>                                   
                                    <action application="set" data="hangup_after_bridge=yes"/>
                                    <action application="bridge" data="sofia/internal/${destnumber}@192.168.1.1"/>
                                    <action application="hangup" data="${originate_disposition}"/>
If you need to setup a DID then you could do something like this.
<extension name="to-switchboard" >
                          <condition field="destination_number" expression="xxxxxxxx">  ( REPLACE xxxxxx with the number dialled on the PSTN)
                                    <action application="set" data="span=${sip_h_X-FreeTDM-SpanName}"/>
                                    <action application="log" data="INFO Incoming CALL for switchboard on span=${span}"/>
                                    <action application="set" data="hangup_after_bridge=yes"/>
                                    <action application="bridge" data="sofia/internal/yyyyyyyyy@192.168.1.1"/>   (REPLACE yyyyyy with the destination number of the SIP Phone this call needs to go to)
                                    <action application="hangup" data="${originate_disposition}"/>
                          </condition>
            </extension>
If your calls need to be routed through the WAN to branch locations etc and you don't want to use the default ulaw/alaw then you could force the codec on the B-Leg of the call to what ever codec you prefer. This will depend on what codecs is supported on the end devices as well as if your card supports transcoding.
It could look something like this:
<extension name="to-switchboard" >
                           <condition field="destination_number" expression="xxxxxxxx">            
                                    <action application="set" data="span=${sip_h_X-FreeTDM-SpanName}"/>
                                    <action application="log" data="INFO Incoming CALL for switchboard on span=${span}"/>
                                    <action application="set" data="hangup_after_bridge=yes"/>
                                    <action application="bridge" data="sofia/internal/yyyyyyyyy@192.168.1.1"/>
                                    <action application="hangup" data="{absolute_codec_string=GSM} ${originate_disposition}"/>
                                    <action application="hangup" data="${originate_disposition}"/>
                           </condition>
            </extension>
These are just simple examples you can use.
Hope this is helpful.
Regards
Deon
On Aug 17, 2011, at 5:32 PM, Ben Naylor wrote:
Hello
Apologies if this is in the wrong place, I am a complete beginner at Freeswitch!
Has anyone had much joy with using the above to route calls from ISDN to a SIP provider?  I have setup the SMG which is a stripped down version of Freeswitch, but am struggling to work out what to configure to get this working.
So far I have tried to set up an external gateway to my provider, but this hasn’t appeared in the SMG Web-gui as a SIP profile.  My provider doesn’t required auth by the way, they have just given me an IP to connect to.
Any help is greatly appreciated
Kind regards
Ben
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