Re: [Freeswitch-users] Sangoma Media Gateway

20 Aug 2011

      Hi Deon

Did you encounter any problems when editing the dialplan?

I'm not 100% sure, but it would appear that whenever I try and edit
default.xml dialplan, the SMG changes status to 'down' in the webgui.  I
then change it straight back again and it comes back up briefly.

However, I've got to the point now where I can't seem to get the SMG to work
again, the TDM drivers start fine, but the SMG just does not want to start.

I have also check the sangomagw log file in /usr/local/smg/log/ but this
seems to stop logging anything once the SMG goes down.

Do you know of any other log files I could check?

Also, if I were to re-install the SMG, would re-running the make/install
commands just re-install over the top?  As a newbie to linux I am still not
sure if this is the case! J

Thanks for your help.

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/${destnumber}@192.168.1.1
<mailto:sofia/internal/$%7bdestnumber%7d@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/${destnumber}@192.168.1.1
<mailto:sofia/internal/$%7bdestnumber%7d@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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