- FreeSWITCH-users - FreeSWITCH Mailing Lists

VPX on windows?
by Alexander Haugg 01 Apr '17

01 Apr '17

Hi, in the newest main brunch I can see the vpxlib project, but after the build I did not found the libvpx.dll or a mod_vpx.dll. Is the VPX video support on windows available? Thanks a lot!

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possible to use mod_fifo without parking a caller?
by Dave Horton 01 Apr '17

01 Apr '17

I’d like to use something like mod_fifo, if possible to support my use case: - caller calls in - freeswitch sends 183 (pre_answer) but does _not_ answer the call - via out of bands means, I send information to an “agent” that a caller is waiting (providing uuid, or parking slot) - agent calls in (to the “parking slot” or equivalent) - caller is now bridged with agent - OR, if no agent calls in within X seconds, caller is rejected with a SIP 480 Is something like that possible with mod_fifo, or other module? Dave

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DTMF games for FreeSWITCH
by Michael Nielsen 31 Mar '17

31 Mar '17

I'm looking for play a bit with FreeSWITCH/SIP and DTMF games. Does anyone know of any free one source games for FreeSWITCH which can be played without a GUI, but via the phone?

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something like transferring to an extension when caller abandons?
by Dave Horton 30 Mar '17

30 Mar '17

I’m hoping to find a dialplan-based solution to this problem. I have a call coming in, and I start executing a dialplan with various applications. At some point, if the caller cancels before I have bridged or answered his or her call, I would like the ability to immediately execute some code — e.g. by jumping to another extension (or something like that) at the moment when the caller abandons. I know this could be orchestrated in an event socket app, but as mentioned I would prefer a dialplan-based approach, if possible. Any ideas? Dave

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Executing java application within an HTTAPI application
by Thomas, Tom 30 Mar '17

30 Mar '17

Greetings everyone, I was wondering if mod_java is supported by the "execute" work action in HTTAPI. So something like this within a response document: <document type="text/freeswitch-httapi"> <work> <execute application="java" data="{args_for_java_app}" ..../> </work> </document> I haven't setup a webserver for HTTAPI yet. So I wanted to know before I go down this path that this can be done. I have a java application that I currently execute from the dialplan. I need to know if httapi can use execute to do the same. Thanks a lot in advance.

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mod_conference ghost and outcall flags
by David Ponzone 30 Mar '17

30 Mar '17

There is obsolete information about conference_auto_outcall_flags in Confluence. This variable is now called conference_utils_auto_outcall_flags. I wanted to update it but I don’t have the rights as I used to have on the old wiki. If anyone can do it… Also, I tested the ghost flag on a conf member and the member is then mute/deaf. For sure, it’s a total ghost :) I used to think ghost was not related to mute/deaf but only to the conference members count. Am I missing something ? David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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How do you install the AMQP module?
by Hinchy, Martin 29 Mar '17

29 Mar '17

Hi guys, I have just installed FreeSWITCH on a Windows 7 x64 PC using the Windows msi 64-bit installer. The installation worked fine and I have been able to make calls through the switch. It appears though that the AMQP module (mod_amqp.dll) is not included in a standard install. It does not appear in the C:\Program Files\FreeSWITCH\mod folder. Does anyone know where I can get this module from? Regards Martin Queensland Rail's number one priority is safety. Queensland Rail has a program of testing for alcohol and illicit substances. Please be aware any person who enters a Queensland Rail premises for business purposes may be randomly tested for alcohol (breath test) and illicit substances (oral fluids test). Please also ensure you are wearing the appropriate Personal Protective Equipment for the site you are visiting. Call your Queensland Rail contact person if you need more information. This email (which includes all attachments and linked documents) is intended for and is confidential to the addressee; it may also be subject to legal professional privilege or otherwise protected from disclosure. If the addressee is a government agency in receipt of a Right to Information Act (2009) application in relation to this email, contact must be made with Queensland Rail ABN 68 598 268 528 in accordance with the third party consultation process provided for in Part 3, Division 3, Section 37 of that legislation. If you are not the addressee, or if you have received this email in error, you must not use, rely upon, disclose or reproduce it (or any part of it) in any way. Please notify the sender of your receipt of it and delete it in its entirety. Neither Queensland Rail (or any of its related entities) accepts any liability for computer viruses, data corruption, delay, interference, interception, unauthorised access or amendment of this email. The views expressed in this email, unless clearly stated otherwise, are the views of the sender. They do not necessarily represent the view or policy of Queensland Rail or any of its related entities.

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Freeswitch not offering VP8
by Jack Loranger 29 Mar '17

29 Mar '17

We are running FreeSwitch on a 2012 windows server. Trying to get Video conferencing to work with verto. Freeswitch is not offering VP8, when I do a "show codec" it is not listed. I read that it is in the core now. libvpx is compiling alright. Is there something I have to do to activate it? Here is vars.xml configuration: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM,G722,7221@32000h,G7221@16000h,OPUS,VP8,H264"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,GSM,PCMA,OPUS,VP8,H264"/> Thanks for your help! Jack

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CoreSession::hangup after MOVED on IVR calls
by Piccinin Francesco 29 Mar '17

29 Mar '17

Hi Brian, many thanks for your feedback! Find below some extract from our FS dialplan. Our SIP system is composed by an Opensips front-end that balances calls on Freeswitch hosts. Realtime data and static config are backended on mysql db. default.xml: a lua script profiling caller / called setting channel_variables. services.xml: <include> <condition field="${service_type}" expression="IVR" break="on-true"> <action application="set" data="default_language=it"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="force_transfer_context=ivrs"/> <action application="lua" data="services/ivr.lua ${did}" inline="true"/> <action application="answer"/> <action application="ivr" data="${ivr_name}"/> <action application="transfer" data="${fallback}"/> </condition> </include> calls.xml: <include> <condition field="${service_type}" expression="^NONE$" break="on-true"> <action application="info"/> <action application="lua" data="functions/consult_xfer_limit_int.lua ${ext} ${cCalled} ${domain_name} ${maxcalls} ${busytriggerA} ${shd_line} ${maxcallsB} ${busytriggerB}"/> <action application="set" data="effective_caller_id_number=${ext}"/> <action application="set" data="effective_caller_id_name=${my_dname}"/> <action application="set" data="hold_music=${moh_path}" inline="true"/> <action application="set" data="transfer_ringback=${moh_path}"/> <action application="export" data="pbx=${pbx}"/> <action application="export" data="domain_name=${domain_name}"/> <action application="export" data="maxcalls=${maxcalls}"/> <action application="limit" data="db ${domain_name} BSY_${cCalled} ${busytriggerB} !USER_BUSY"/> <action application="limit" data="db ${domain_name} MAX_${ext} ${maxcalls} !USER_BUSY"/> <action application="limit" data="db ${domain_name} BSY_${ext} ${busytriggerA} !USER_BUSY"/>--> <action application="bind_meta_app" data="1 ab s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 ab s execute_extension::att_xfer XML features"/> <action application="bind_meta_app" data="3 ab s execute_extension::parking XML features"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="export" data="alert_info=http://127.0.0.1/ring.wav;info=${alert_info};x-line-id=0"/> <action application="set" data="bypass_media=${lan}"/> <action application="set" data="t38_passthru=true"/> <action application="export" data="t38_passthru=true"/> <action application="set" data="disable_q850_reason=true"/> <action application="set" data="sip_redirect_context=redir"/> <action application="export" data="sip_redirect_context=redir"/> <action application="export" data="nolocal:execute_on_answer=lua functions/clean_pickup.lua ${uuid}"/> <action application="set" data="callee_id_name=${cid_name}"/> <action application="export" data="callee_id_name=${cid_name}"/> <action application="set_profile_var" data="callee_id_name=${cid_name}"/> <action application="set" data="leg=A"/> <action application="export" data="nolocal:leg=B"/> <action application="bridge" data="{sip_invite_domain=${domain_name},sip_cid_type=pid,presence_id=${cCalled}@${domain_name}}${sofia_contact(${cCalled}@${domain_name})}"/> <action application="set" data="continue_on_fail=USER_BUSY,ORIGINATOR_CANCEL"/> <action application="hangup" data="USER_BUSY"/> </condition> </include> redir.xml: <include> <context name="redir">  <extension name="redirections"> <condition field="destination_number" expression="^(\d+)$"> <action application="log" data="INFO REDIRECTED"/> <action application="info"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bypass_media=false"/>  <action application="set" data="leg=A"/> <action application="export" data="nolocal:leg=B"/> <action application="lua" data="functions/call_redirect.lua ${cCalled} ${destination_number}"/> </condition> </extension> </context> </include> call_redirect.lua profile cCalled (forwarder ext) and destination_number (forward destination) check session status with session:ready() and execute a bridge with: app="{ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_name='Da:"..caller.."',originate_timeout=25}sofia/gateway/"..gw[i].."/"..redirected; session:execute("bridge", app); Please note that a-leg is terminated just after b-leg is terminated with originator_cancel (sip 487) on SIP moved received from forwarded extensions. Thanks in advance for you help Francesco

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CoreSession::hangup after MOVED on IVR calls
by Piccinin Francesco 29 Mar '17

29 Mar '17

Hi all, we are experiencing a call drop issue when users configure call forward unconditional on their extensions. Call drops when these extensions are reached from pstn and through an IVR post selection. Freeswitch recived a SIP 302 Moved from these endpoints, call are processed on sip_redirect_context but freeswitch first send a SOFIA DESTROY to forwarding extension (correct) and then KILL A-leg (IVR) with the following error: CoreSession::hangup It seems freeswitch does not keep alive a-leg in this scenario. Can you please help us? Thanks Francesco

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