[Freeswitch-users] T.38 Fax

David Witham david.witham at netsip.com.au
Thu Jul 16 09:32:45 MSD 2015


Hi Dave,

It is not a sofia param it is a channel variable so you should set it in the dialplan. Note that it is t38_passthru not t38-passthru. There should be no performance penalty using this variable.

regards,
David
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Dave Greeko <davegreeko at yahoo.com>
Sent: Thursday, 16 July 2015 11:45
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] T.38 Fax

Many Thanks David. Should I set this once in Sofia Profile (<param name="t38-passthru" value="true"/>) or per dial-plan extension? lastly, if I can do it in the sip profile without worrying about the dial plan will that have any performance cost in terms of system resources - just a question?
P.S: I know for a fact FreeSwitch awesome community may not like these questions because they want us to try it and make it work first than ask....  I'm sorry in advanced


--------------------------------------------
On Wed, 7/15/15, David Witham <david.witham at netsip.com.au> wrote:

 Subject: Re: [Freeswitch-users] T.38 Fax
 To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
 Date: Wednesday, July 15, 2015, 5:50 PM

 Hi Dave,

 You need to set
 t38_passthru=true in your dialplan to allow the t38 reINVITE
 to traverse your FS.

 regards,
 David
 ________________________________________
 From:
 freeswitch-users-bounces at lists.freeswitch.org
 <freeswitch-users-bounces at lists.freeswitch.org>
 on behalf of Dave Greeko <davegreeko at yahoo.com>
 Sent: Thursday, 16 July 2015 08:50
 To: freeswitch-users at lists.freeswitch.org
 Subject: [Freeswitch-users] T.38 Fax

 Hello All,
 I'm trying to just relay t.38 fax calls
 along the voice calls between two endpoints with FS in the
 middle.
 In the dial-plan I set the proxy
 media mode to true which works extremely well with my setup
 considering the codec negotiation headache for my voice
 calls. My question is do I have to explicitly set any Sofila
 parameters in the SIP profile or any variables related to
 T.38 in the Bridge dial string to make the fax go from
 Caller to Callee through FS?

 My Freeswitch version is 1.4.19 and mod_spansdp
 is already loaded

 Regards,
 Dave

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