Hi Michael, I am getting nothing out of that. On Fri, Mar 29, 2013 at 1:55 AM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: Access to rxfax channel variables (Michael Collins) 2. Re: callee id inbound (Steven Schoch) 3. Re: mod_perl hangupCause() returning 'NONE' and hangup_cause is null (Michael Collins) 4. Re: Calling extension using ruby (Michael Collins) 5. Re: callee id inbound (Hermouet Erwan)
---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 28 Mar 2013 13:17:20 -0700 Subject: Re: [Freeswitch-users] Access to rxfax channel variables http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook
-MC
On Thu, Mar 28, 2013 at 2:33 AM, Eugene Prokopiev <enp@itx.ru> wrote:
Hi,
How to read rxfax channel variables described in http://wiki.freeswitch.org/wiki/Mod_spandsp#Controlling_the_app?
My dialplan:
<action application="set" data="api_hangup_hook=log INFO FAX received from ${caller_id_number}(${fax_remote_station_id}) : ${fax_result_text}"/> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="rxfax"
data="/var/spool/fax/$1/${strftime(%Y%m%d%H%M%S)}-${caller_id_number}.tif"/> <action application="hangup"/>
But I see only:
2013-03-28 13:28:40.256497 [INFO] mod_commands.c:5899 FAX received from 2440663() :
-- Regards, Eugene Prokopiev
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
---------- Forwarded message ---------- From: Steven Schoch <schoch+freeswitch.org@xwin32.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 28 Mar 2013 13:18:18 -0700 Subject: Re: [Freeswitch-users] callee id inbound I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that.
-- Steve
---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 28 Mar 2013 13:18:58 -0700 Subject: Re: [Freeswitch-users] mod_perl hangupCause() returning 'NONE' and hangup_cause is null In a case like this you are better off doing a uuid_dump to a temp file and sifting through the output to see what is actually available. My guess is that there will be something in there that you can use.
-MC
On Thu, Mar 28, 2013 at 4:50 AM, Ashish gautam <ashish@nms.co.in> wrote:
Hi,
I am running a perl script after the call lands on my dialplan. I want to get the status that whether the call was originated properly and and if yes what was the hangup cause. I am getting the hangup status using hangupCause() and it returns 'NONE' everytime no matter what the reason is. Also the hangup_cause variable is nill or not set everytime.
Any help is appreciated.
Thanks in advance!!
-- Regards
Ashish Gautam
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 28 Mar 2013 13:22:51 -0700 Subject: Re: [Freeswitch-users] Calling extension using ruby something like this:
originate {ignore_early_media=true}sofia/gateway/gwname/${phone_num} &speak(flite|kal|${text_to_speak})
-MC
On Thu, Mar 28, 2013 at 1:54 AM, Amit Kumar <findme@itsamit.com> wrote:
I am trying to make an application that can initiate a call from the Web app to a PSTN line. So when a user clicks on call, the PSTN line will ring, and FS will play the text entered on the web app to the PSTN.
I am able to dial the PSTN from a SIP phone(Zoiper). I am also able to call the PSTN line from the CLI, but I have no clue as to how to eliminate the SIP Phone all together.
I can connect to FS using ESL, and send the command to initiate a call, but since no user is registered without the SIP phone registering them, I am not sure how to go ahead.
Any help is appreciated!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
---------- Forwarded message ---------- From: Hermouet Erwan <ehermouet@bluetel.fr> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 28 Mar 2013 21:25:25 +0100 Subject: Re: [Freeswitch-users] callee id inbound Yes it s that
Steven Schoch <schoch+freeswitch.org@xwin32.com> a écrit :
I understand. I'm sure there must be a parameter you can put in the "user/12345@${domain} action, but I don't know what that is. Someone more knowledgeable will have to answer that.
-- Steve
------------------------------
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Hermouet Erwan Responsable technique Bluetel
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