March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Re: [Freeswitch-users] UK ring tone
by Michael S Collins 13 Mar '10

13 Mar '10

You can set it in the dialplan prior to transferring or bridging the incoming call. Sent from my iPhone On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry(a)gmail.com> wrote: > Hi, > > I've checked the internal and external sip profiles and think this > needs to be set in autoload switch file as per: > > http://wiki.freeswitch.org/wiki/Variable_transfer_ringback > > I want by default a UK ringtone to be heard when I call a DDI that is > configured on the system. At the moment it sounds like I'm calling the > US. > > Thanks. > > -- http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org

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Gateway context for incoming calls
by Tim St. Pierre 13 Mar '10

13 Mar '10

Hello, Is there a way for incoming calls from a gateway to go to a context other than the default one set in the sofia profile? I need a way to separate calls from providers vs. calls from the public at large. Here's the scenario: Our external sofia profile maps to the public IP address and listens on port 5060. Our providers send calls to this address, but we also need to have anonymous sip calls come in to the public address, on the default port. We want the calls from our SIP trunks to get billed appropriately, and increment our mod_limit counters, etc. If a call comes in from somewhere other than a SIP trunk, we want it to have access to extensions, but be rated as a free call. All our SIP trunks are mapped using static addresses - there isn't any registration or authentication happening in either direction. Each one has a gateway entry. Thanks! -Tim

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voicemail_greeting_number stickiness
by Tim St. Pierre 13 Mar '10

13 Mar '10

Hello all, I'm trying to set up our voice mail system so that greetings can be mapped to failure conditions, ie. "I'm in my office, but not taking calls" when a phone is busy or in DND mode, but "Sorry I'm not here" if no_answer. I programmed things to set voicemail_greeting_number in the dialplan based on the bridge failure cause set when we rang the phone, but when I execute the voicemail application, it stores that greeting number in the database. I would much rather override for this call only, so that the user can still take advantage of the voicemail application's option to select a greeting. (which would be used as the no answer greeting under normal circumstances). Any thoughts? -Tim

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OpenZap/Sangoma drivers
by Mark Sobkow 12 Mar '10

12 Mar '10

I filed a bug report this morning because of a Freeswitch build problem, but the resolution was to update the OpenZap/Sangoma drivers. Updating the Sangoma drivers was a little hairy because of our project build process needing to be updated, but I got it figured out and got the latest build of Freeswitch installed (svn checkout this morning.) Next I need to re-test the call recordings.

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DTMF problem
by Sean Holt 12 Mar '10

12 Mar '10

Question: I have FS 1.0.5 installed on a Centos 5.4 server box. I hope there¹s a simple answer to this question. I¹m having problems in particular when connecting to a outside conference line on a completely different phone system entering numbers that the receiver should accept. For example, call into a conference line, auto-attendent ask for the pin number, I then enter pin number, but The receiving ivr tells me I¹ve entered the wrong pin number. I¹m assuming that I¹ve reached a timeout and the ivr is letting me know that I¹ve entered the incorrect pin number. My thinking is that this is a DTMF issue. I¹m new to FS so not sure if I need to enable something in my dialplan or not. Thoughts? Thanks in advance Sean

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Multiple gateways dial string and user busy
by Max Ivanov 12 Mar '10

12 Mar '10

Hi all! when I do test call from fs_cli: originate sofia/gateway/panas110/223|sofia/gateway/panas111/223 &playaback(local_stream://moh) If firest attempt returns USER_BUSY it tries to call via second one. Is it normal? How can I stop calling attempts after first USER_BUSY?

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