June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Simulating v.23 modems
by lists＠collisters.com 07 Jun '13

07 Jun '13

I know this might seem an odd request, but has anyone ever heard of a modem implementation in software for FreeSWITCH, it occurs to me that it must be possible to answer a call from a modem and decode the data from the RTP packets. I have a remote device that calls in to a v.23 modem, but I would like to find a more modern solution for answering that call that doesn't use old hardware. Thanks in advance Paul C.

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ESL client managed Windows/x64 build
by Rafal Gwizdala 07 Jun '13

07 Jun '13

Hi, does anyone have a complete x64 ESL managed client build (including x64 ESL.dll library)? Could you share it with me? I have some problems building a 64 bit version of Freeswitch... Thanks RG

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Fwd: Originating a call between an user and mod_httapi-based IVR service - passing some correlation ID
by Rafal Gwizdala 07 Jun '13

07 Jun '13

Hi, i'm just beginning to learn FreeSwitch and have a quick question to you I want to implement an automated IVR-like service that will be used for calling selected customers and performing a survey/collecting some data from them. I decided to use mod_httapi for handling the calls, but since my application will be calling the customers I need to initiate the call too. I'd like to do that using an inbound event socket, by sending 'originate' command. And my questions are: 1. Is this a correct approach, or maybe there's a better method of initiating such call? 2. How to pass some correlation id when originating the call so that it would be delivered back to my application with mod_httapi request? I just need to correlate the initiation of the call with the call handler code that will be invoked later. Thanks and best regards Rafal

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Setting up callbacks
by Alex Lake 07 Jun '13

07 Jun '13

I was wondering where's a good place to find some examples of how, by sending the right commands to the event_socket, I could have Freeswitch establish callbacks for me. Essentially there are a couple of different termination types - PSTN (via a gateway) and internally registered SIP accounts I would wish to be able to set up A->B (and maybe small conferences) using these types of destination in any combination. I've looked through the event_socket pages and the "originate" syntax, but would like to put together something a little more "idiot-friendly" so am looking around for precedents/tips...

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Freeswitch Java ESL Client
by Thiruvallar Raveendran 07 Jun '13

07 Jun '13

Hi All, I am new to FreeSWITCH. I am using Java ESL client to receive call events from FreeSWITCH machine. Memory leak occured when everytime Freeswitch connects with my java code through ESL client. Due to some reason, garbage collector doesn't clean up the memory. My Java program fails without memory when I feed thousands of calls. I cleared all my local references and closed netty channel. I am not sure whether ESL clients holds any reference. Is anyone faced this issue before? Your help on this will be much appreciated. Thanks. Regards, Thiru

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FreeSwitch IVR
by Vishal Kakkar 07 Jun '13

07 Jun '13

HI all, I am working on Freeswitch IVR since long time now.. But have been creating xml scripts manually for IVRs. Can you guys please share is there any GUI tool (Open source) available where in we can drag and drop to create XML scripts? Thanks, -Manav

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recording one leg of the call in vm after an agent in fifo answered the call
by Jason Holden 07 Jun '13

07 Jun '13

Any idea why this would happen?

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Need to build FAX service
by pratham2013 07 Jun '13

07 Jun '13

Hello, I need to build the FAX service which can send can send and receive through email so is it possible to build with FreeSwitch? Do I need to purchase any 3rd party like SIP service provide which can used for FAX? if is that any then can we any one provide example for the same? Please help me in this matter. Thanks Manish Khilwani -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-to-build-FAX-service-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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how to pass session from dialplan xml to lua script?
by Vincent Xia 07 Jun '13

07 Jun '13

hi all, i have a lua script(record.lua) to record the call: record.lua -------------------------------------- callerid = argv[1] dest = argv[2] recording_dir = argv[3] filename = callerid ..dest recording_filename = string.format('%s%s', recording_dir, filename) if session:ready() then -- NOTE: the session here is nil! max_len_secs = 30 silence_threshold = 30 silence_secs = 5 test = session:recordFile(recording_filename, max_len_secs, silence_threshold, silence_secs); end but i don't know how to pass "session" from the default dialplan to lua in the following sentence: <action application="lua" data="record.lua ${caller_id_number} ${destination_number} $${recordings_dir}"/> any response is apprieciated!

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Reinvite - Codec G.729 decoder error
by Kevin Mathy 07 Jun '13

07 Jun '13

Hi list, I have an issue in my Freeswich in SBC mode. It's a call from Freeswitch to my endpoint, my endpoint answer G729, after this 200OK, my end-point re-invite in G711A, and my freeswitch say: "Codec G.729 decoder error". Ex : 1) SIP Provider ----INVITE(PCMA, G729)-----> FS ----INVITE(PCMA, G729)-----> End-point 2) SIP Provider <-----(200 OK G729)------ FS <------(200OK G729)------ End-point 3) FS <------(Re-INVITE PCMA)------ End-point 4) FS Error "Codec G.729 decoder error" in logs In fact, it's something like http://jira.freeswitch.org/browse/FS-3739, but the patch doesn't seem to work in our situation. My freeswitch's conf: *sip_profile TOTO: * * * * <param name="dtmf-type" value="rfc2833"/>* * * * * * <param name="rfc2833-pt" value="101"/>* * * * <param name="inbound-late-negotiation" value="true"/>* * * * <param name="inbound-codec-prefs" value="PCMA,G729,PCMU"/>* * * * <param name="outbound-codec-prefs" value="PCMA,G729,PCMU"/>* *Diaplan TITI:* *<action application="export" data="codec_string=${ep_codec_string}"/>* *<action application="set" data="hangup_after_bridge=true"/>* * <action application="set" data="ignore_early_media=true"/>* *<action application="set" data="inherit_codec=true"/>* *<action application="export" data="t38_passthru=true"/>* *<action application="export" data="suppress_cng=true"/>* *<action application="bridge" data="* * {* * sip_invite_params=user=phone,* * sip_cid_type=none,* * ignore_display_updates=true,* * sip_renegotiate_codec_on_reinvite=true,* * }* * sofia/gateway/${distributor(LISTTMP ${sofia(profile TITIPRO gwlist down)})}/${numberdest}"/>* * * * * We choosed to use inbound-late-negotation with inherit-codec because of our t38_passthru option (as described here in FS wiki, I don't exactly remember where) To resolve this problem, we tried to find how to make FS forward the Re-INVITE received on B-Leg to the A-Leg, but we didn't found any solution... If someone have an idea, it would be great ! ;-) Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * --

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