Hello again guys, 

a little bit of progress but not yet working fully - couple of further questions - to recap the scenario is to use freeswitch to split fax and voice, and forward on to either remote phone or remote fax accordingly - the below bridge on to remote fax fails as t.38 needs to be supported with a re-invite & SDP and at the moment freeswitch does not offer t.38 in its 200 ok to the re-invite, log suggest no update to SDP - so not clear how to ensure its an end to end update with t.38 set up correctly,  I had thought that ""export" data="t38_passthru=true"/>" would help, but has not yet - perhaps I need to apply as global setting? Any further thoughts  ?

    <extension name="fax_test">
    <condition field="destination_number" expression="(\+)?4412345678$">
    <action application="export" data="t38_passthru=true"/>
    <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default"/>
    <action application="sleep" data="1000"/>
    <action application="transfer" data="1008 XML default"/>
    </condition>

    </extension>
    <extension name="fax">
    <condition field="destination_number" expression="fax">
    <action application="export" data="t38_passthru=true"/>
    <action application="bridge" data="{sip_renegotiate_codec_on_reinvite=true}sofia/gateway/hgw/+4488888888888"/>
    </condition>
    </extension>

The Cli shows as -
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4276 Audio Codec Compare [CLEARMODE:121:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4276 Audio Codec Compare [t38:125:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4276 Audio Codec Compare [t38:125:8000:20:0:1]/[opus:116:48000:20:0:1]
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4276 Audio Codec Compare [t38:125:8000:20:0:1]/[G722:9:8000:20:64000:1]
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4276 Audio Codec Compare [t38:125:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4567 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event.
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:4596 sofia/external/+4488888888888 Set 2833 dtmf send payload to 101 recv payload to 101
2016-10-04 12:59:56.552337 [DEBUG] switch_core_media.c:6417 Audio params are unchanged for sofia/external/+444488888888888.
2016-10-04 12:59:56.552337 [DEBUG] sofia.c:7783 Processing updated SDP
2016-10-04 12:59:56.552337 [DEBUG] sofia.c:6962 Channel sofia/external/+444488888888888 entering state [completed][200]
2016-10-04 12:59:56.572337 [DEBUG] sofia.c:6962 Channel sofia/external/+444488888888888 entering state [ready][200]
2016-10-04 12:59:56.572337 [NOTICE] sofia.c:1011 Hangup sofia/external/+444488888888888 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2016-10-04 12:59:56.572337 [DEBUG] switch_ivr_bridge.c:712 sofia/external/+444488888888888 ending bridge by request from read function
2016-10-04 12:59:56.572337 [DEBUG] switch_ivr_bridge.c:785 BRIDGE THREAD DONE [sofia/external/+444488888888888]

Appreciate any tips, or am I best locally capturing the fax, as in store and forward as opposed to trying to forward fax call?

thanks
Magnus
 

On 20 September 2016 at 23:59, Brian West <brian@freeswitch.org> wrote:
<action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" /> 

Attaches a media bug, This listens for the fax tone and proceeds into the dialplan, If it detects a fax tone in the first 5 seconds (5000ms) it transfers to the extension fax XML default, which is fax extension in the xml dialplan, default context.

I think you may be over thinking it slightly.

/b


On Tue, Sep 20, 2016 at 5:25 PM, Magnus Kelly <magnus.kelly@gmail.com> wrote:
Appreciate the url form both Brian & Don, I had found this on Confluence,  but I read it as more about how to detect a fax and send it to fax extension – that I get. Hence am I thus to understand that the logic would be if not triggering this transfer to the fax extension;  to let it “fall through” to the next action and send it to a normal user extension? 

Assuming correct, is the voice caller experience acceptable in terms of timing for a human (impatient as ever)? Roughly whats the minimum time needed to detect a remote fax? Can ring back be played out in parallel to the fax detection stage ? I will try myself but I do not yet have the fax machines to experiment with, hence asking for advice. 
Thanks

From: <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Brian West <brian@freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Date: Tuesday, 20 September 2016 at 23:00
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Fax / voice splitting question


On Tue, Sep 20, 2016 at 4:34 PM, Magnus Kelly <magnus.kelly@gmail.com> wrote:
Hello all ,

Please could someone share if its possible to split voice and fax calls, e.g. numbers that are used to also receive fax’s to be forwarded by SIP to FreeSwitch in order to answer calls and then somehow automatically determine between voice and fax calls and onward route to either fax extension or to a hunt group to find a person? I was thinking of looking for T.38 in the SDP, but then understood that this only appears in the actual fax transaction and thus bit late in the setup to determine routing.

Appreciate any tips if others have successfully found solutions to this call case.

Thanks
Magnus

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