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mod_nibblebill and dynamic conference
by Alfredo Quiroga-Villamil 14 Mar '10

14 Mar '10

All: We are currently exploring the possibility of using mod_nibblebill and integrating it with a conference. It seems like it offers all the features and more that we need. The way we create a conference now is as follows: originate sofia/gateway/url/XXXXXXXXXXX &conference(myConf+flags{moderator,wait-mod}) >From the mod_nibblebill wiki page it seems like we need to set a few variables for the B leg: enable_heartbeat_events nibble_rate nibble_account= I was wondering if there is a way to set these variables via the API call shown above where the conference is created. If not, what options do you recommend to accomplish this? Thanks in advance, Alfredo

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Trouble with P-Asserted-Identity header(s)
by Roland Hänel 14 Mar '10

14 Mar '10

Hello, I'm developing a very simple call-through application using FreeSWITCH, but currently I'm stuck because my provider requires me to send a 'P-Asserted-Identity' header for outgoing calls if I want so signal an arbitrary caller ID in the 'From' field. Setup is as follows: - a-leg comes in from an external gateway - dialplan dispatches the call to a perl handler script - perl script looks like this (simplified as much as possible for this description): $num = "02222222"; $session->answer(); $session->setVariable("effective_caller_id_name", "The Redirector"); $session->setVariable("sip_h_P-Asserted-Identity", '< sip:0123456789@q-loud.net <sip%3A0123456789(a)q-loud.net>>'); $session->execute("bridge", "sofia/gateway/mygateway/$num); The problem is now that FreeSWITCH correctly insertes a P-Asserted-Identity header as set in the perl script. But there is already an existing P-Asserted-Identity header by default, so I end up with two of them. This is the outgoing INVITE (b-leg): INVITE sip:02222222@x.x.x.x SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N Max-Forwards: 63 From: "The Redirector" <sip:01111111@q-loud.net <sip%3A01111111(a)q-loud.net> >;tag=DBSvZ9NZc750g To: <sip:02222222@x.x.x.x> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02 CSeq: 128071597 INVITE Contact: <sip:gw+mygateway@x.x.x.x:5080;transport=udp;gw=mygateway> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 244 X-Port: 5060 X-FS-Support: update_display P-Asserted-Identity: "The Redirector" <sip:01111111@q-loud.net<sip%3A01111111(a)q-loud.net> > P-Asserted-Identity: <sip:0123456789@q-loud.net<sip%3A0123456789(a)q-loud.net> > [... sdp ...] Numbers are: 01111111 - the original caller (a-leg & b-leg From:) 02222222 - the number redirected to (b-leg To:) 0123456789 - the number my provider needs in P-Asserted-Identity ('redirector's number') So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789 line, but unfortunately it already includes another P-Asserted-Identity 011111111 line, and that breaks the setup at my provider. I really appreciate any help. Greetings, Roland -- Roland Haenel QSC AG - http://www.qsc.de

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Call transfer with Polycom works ok but a call is left on hold?
by Yehavi Bourvine 14 Mar '10

14 Mar '10

Hello, We have a strange behavior which my users told me today, but I have a feeling it lasts for a long time (we are running now version 16972). User A calls user B on a Polycom. User B press the "transfer" key and calls C who answers. B then press "transfer" again, and now A talks to C. However, B is left with a held call to the voicemail of C... Anyone saw this behaviour? unattended transfer and conference work ok. Thanks! __Yehavi:

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how to access session variables after hangup???
by rentmycoder rentmycoder 14 Mar '10

14 Mar '10

Hi guys, Freeswitch rocks! except the documentation:( I'm not able to figure out how to access session variables after hangup... I try to dial out on a gateway and detect it the called party hangs up before bridge... bgapi originate {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun hangup.lua ${uuid}'}sofia/gateway/phoneno &park() I cannot access the variables using: 1. script arguments: 'luarun hangup.lua ${uuid}'} 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data (${api_hangup_hook} contains a variable) 2. session:getVariable("originate_disposition") 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt to call global 'getVar' (a nil value) 3. params:getHeader("variable_sip_req_uri") 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt to index global 'params' (a nil value) I have asked this before on the dev list too, but Anthony didn't gave me a clean answer here: http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html "In the script you can access the data from the env event which is automatically created for you and it contains all the channel variables from that departed channel." Ok, but how????? Please help...

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js.dll crash on Windows Server 2008 R2 x64
by Quentin CALVEZ 14 Mar '10

14 Mar '10

Hello everybody, I've been trying Freeswitch those last days and there is one error that I don't know what to do about. I'm using FusionPBX to manage my freeswitch installation and it works great, however, whenever I try to use a function involving javascript (auto atendants, hunt groups...) the server crashs telling me that the faulty module is js32.dll. I really don't know what to do about it. Could someone help me :) ? Regards. Quentin Calvez

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Inline and cond
by Dome Charoenyost 14 Mar '10

14 Mar '10

I try to use cond in my dialplan <condition field="destination_number" expression="^(1111)"> <action application="answer"/> <action inline="true" application="set" data="b=25"/> </condition> <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$"> <action application="log" data="ALERT ${b} is indeed greater than 12"/> <anti-action application="log" data="ALERT ${b} is not greater than 12"/> </condition> it's work but when use set b=${b1} it's not work <condition field="destination_number" expression="^(1111)"> <action application="answer"/> <action inline="true" application="set" data="b1=25"/> <action inline="true" application="set" data="b=${b1}"/> </condition> <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$"> <action application="log" data="ALERT ${b} is indeed greater than 12"/> <anti-action application="log" data="ALERT ${b} is not greater than 12"/> </condition> What's wrong ? inline or cond ? Dome C.

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conferences lead to high server load
by Christian Löschenkohl 14 Mar '10

14 Mar '10

hi we do experience a unusual high server load with the latest freeswitch versions. about 50 conference users lead to a server load of over 10 - reproducible by the way. this wans't the case until my latest trunk update. fs version: 16714 os: debian lenny x86_64 has something substantially changed in mod_conference recently? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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internal/external profiles
by Brian May 14 Mar '10

14 Mar '10

Hello, Why is it recommended to use separate profiles for internal and external SIP? This page: <http://wiki.freeswitch.org/wiki/External_profile> suggests it is because of NAT. However this page recommends using separate profiles even if NAT is not an issue: <http://wiki.freeswitch.org/wiki/Auto_NAT>: "NOTE: It is still recommended that you use a second profile for your SIP providers. The default conf/sip_profiles/external.xml is set up specifically for use with providers." However I am still left uncertain what this means. Not trying to criticize here, just trying to learn. Thanks. -- Brian May <brian(a)microcomaustralia.com.au>

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Possible typo on Openzap.conf wiki page specifying fxs/fxo-channels
by Robert Hadley 14 Mar '10

14 Mar '10

On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf example concerning specifying the fxo-channel vs. fxs-channel? In the [span wanpipe FXS] section the channels are shown on wiki page as fxo-channels => 1:1 and 1:2 In the [span wanpipe FXO] section the channels are shown on wiki page as fxs-channels => 1:3 and 1:4 Are specifying the fxs-channels and fxo-channels shown in the wrong sections? I had to specify fxs-channels in the FXS span and fxo-channels in the FXO span to get it working on my hardware. Except from wiki page: Sangoma A200/A400 * A200, A200D, A400, A400D series and variants The configuration depends on whether wanpipe is configured to use Zaptel TDM Voice, or the Sangoma standalone TDM Voice API. This is determined in the installation and configuration of the Sangoma wanpipe software. If wanpipe is using Zaptel, you need to configure openzap.conf with [span zt] entries. For example, if ports 1 and 2 are FXS (e.g. configured to accept analog phone connections), and ports 3 and 4 are FXO (e.g. configured to accept PSTN analog lines) you will need: /usr/local/freeswitch/conf/openzap.conf: [span zt FXO] name => OpenZAP number => 3001 fxo-channel => 1 number => 3002 fxo-channel => 2 [span zt FXS] name => OpenZAP number => 4165551111 fxs-channel => 3 number => 4165552222 fxs-channel => 4 If wanpipe is standalone, you need to configure openzap.conf with [span wanpipe] entries. For the same example as above this would be: /usr/local/freeswitch/conf/openzap.conf: [span wanpipe FXS] # This is the value of the callerid_name variable that is raised in the dialplan name => Analog Phone 1 # This is the value of the callerid_number variable that is raised in the dialplan number => 3001 fxo-channel => 1:1 name => Analog Phone 2 number => 3002 fxo-channel => 1:2 [span wanpipe FXO] # the chan_name variable will raised as "OpenZAP/2:1/4165551111" in dialplan when an incoming call arrives on this port name => PSTN line 1 number => 4165551111 fxs-channel => 1:3 name => PSTN line 2 number => 4165552222 fxs-channel => 1:4

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Lua Script with mod_vmd, setInputCallback doesn't seem to get called
by Michael De Lorenzo 14 Mar '10

14 Mar '10

I've got the following Lua script working in a sense, but the InputCallback never seems to get called while the file is being streamed to the call recipient. I've tried moving the "vmd start" command and set input callback around a bit, but to no avail. I'm testing this against a cell phone voice mailbox (Verizon). freeswitch.consoleLog("info","########################################################\n\n"); > > number_to_call = argv[1] > message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2] > > voicemail_detected = false; > > function onInput(s, type, obj) > freeswitch.consoleLog("notice","*********** Type?: " .. type .. " > *************\n"); > -- freeswitch.consoleLog("notice","*********** VMD?: " .. > session:getVariable("vmd_detect") .. " *************\n"); > > if(type == "event" and voicemail_detected == false) then > freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING > MACHINE DETECTED *************\n"); > voicemail_detected = true; > return "break"; > end > end > > function playbackMessage() > sleep_time = 1000; > if(voicemail_detected) then > sleep_time = 2500; > end > -- sleep a second > session:sleep(sleep_time); > -- play a file > session:streamFile(message_to_play); > end > > function notify() > session = > freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1" > .. number_to_call) > > > > if(session:ready()) then > -- answer the call > session:answer(); > session:setInputCallback("onInput", ""); > session:execute("vmd","start"); > > playbackMessage(); > if(voicemail_detected) then > freeswitch.consoleLog("notice","************ DOING PLAYBACK FOR > VOICEMAIL/ANSWERING MACHINE *************\n"); > playbackMessage(); > end > > freeswitch.consoleLog("notice", "********* hanging up session > **********\n"); > -- hangup > session:hangup(); > end > end > > notify(); > > > freeswitch.consoleLog("info","########################################################\n\n"); > -- Michael De Lorenzo

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