March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

conference dialplan with perl scripting
by deniro 21 Mar '11

21 Mar '11

Hi All, I'd like to run perl script in dialplan by passing PIN to it <action application="perl" data="/path/to/your/script.pl $PIN" /> this script is assumed to compare the PIN with the values in database and if there is a match , let go to the conference room (with the perl script I am able to compare database values and determine if there is match for the PIN by manually running it, and I'd like to run this through dialplan ) so, In the dial plan 1--Where exactly I call this perl script after reading PIN with "play_and_get_digits" 2-- what should I return from the perl script if there is match or there is no match for the PIN and pass it to dialplan 3-- if there is a match, how should I proceed to conference 4-- if there is no match how should I terminate(exit) from dial plan Any guidance, links, documents well appreciated thans deniro--

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Outbound dialing to SPA3102 behind NAT
by Nick Rosier 20 Mar '11

20 Mar '11

Hi, I've currently got an Asterisk server running at home but want to switch to FS on my external server (located in a DC). Configuration: - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1) - Couple of voip phones The Asterisk server is running on my local network as are all the other devices; everything is connected to a DSL-router with NAT. The SPA3102 are configured as Trunks in Asterisk with the following configuration: allow=ulaw canreinvite=no context=from-trunk disallow=all dtmfmode=rfc2833 host=dynamic incominglimit=1 nat=never port=5061 qualify=yes secret=trunk1 username=trunk1 type=friend The SPA3102 are configured to register their PSTN-line to the Asterisk server. Outbound routes are configured to use 1 of the 2 Trunks. I want to do the same thing with my FS running on a server in the DC. I've setup all the devices and can make internal calls. Inbound calls on the PSTN lines are also forwarded to the correct voip-devices. I just cannot figure out how to configure outbound dialing to the PSTN-lines. The Wiki (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I should configure them as following: <include> <extension name="To PSTN"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> </condition> </extension> </include> The problem here is that the SPA3102 are behind my router so the only possibility would be to setup some port forwarding to on the router with different ports for the 2 devices. I would rather not do that so I was wondering if there is an easier/better way to do that. Any ideas? Rgds, N.

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Re: [Freeswitch-users] what to do after "make current"
by Philippe Le Toquin 20 Mar '11

20 Mar '11

thanks a lot for the clarification I actually have a init.d script to restart but I will keep your method as well /Philippe On 11-03-20 02:43 PM, msc(a)freeswitch.org wrote: > You can restart freeswitch at your leisure. The new binary won't "hot > swap" by itself. If you want to do it from the Linux cmd line or shell > script you can do something like this: > > fs_cli -x "fsctl shutdown elegant" > sleep 20 > freeswitch -nc -nonat > > Of course, use a sleep value appropriate for the speed of your server > and use the FS cmd line args for your config. > > -MC > > Sent from my HTC on the Now Network from Sprint! > > ----- Reply message ----- > From: "Philippe Le Toquin" <philippe(a)ppmt.org> > Date: Sat, Mar 19, 2011 1:27 pm > Subject: [Freeswitch-users] what to do after "make current" > To: <freeswitch-users(a)lists.freeswitch.org> > > Hello, > > This might sound like a stupid question but I can't find the answer so... > > TO update Freeswitch I do > > git pull > > and then > > make current > > After that I generally restart freeswitch. But do I really need to do > it? or will it somehow manage to "hot swap" with > the new binary? > > If I need to restart. Do I have to do it right after the update or can > it wait until it is night time? > > Thanks

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what to do after "make current"
by Philippe Le Toquin 19 Mar '11

19 Mar '11

Hello, This might sound like a stupid question but I can't find the answer so... TO update Freeswitch I do git pull and then make current After that I generally restart freeswitch. But do I really need to do it? or will it somehow manage to "hot swap" with the new binary? If I need to restart. Do I have to do it right after the update or can it wait until it is night time? Thanks

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Voicemail with no predefined profile
by Moe Navid 18 Mar '11

18 Mar '11

Hi all, I'm in the process of migrating my services from Asterisk to FreeSWITCH. Is there a way to create voicemail boxes dynamically on the fly without predefining user profiles? Right now I'm using OpenSIPS as my registrar and load balancer for my Asterisk boxes. I use asterisk's real time in conjunction with res_config_mysql to I define my voicemail boxes in my mysql table and Asterisk reads it's info from there. Can I do something similar with FreeSWITCH? The only way I can find to do it in FreeSWITCH is using mod_xml_curl. Thank you Moe

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some outbound calls failing
by elijah 18 Mar '11

18 Mar '11

I have put together a dialer (of sorts) that is failing a fraction of the time. First, I instantiate a call leg from FreeSwitch to an agent (an employee in my facility with a softclient registered to FreeSwitch). Upon answering this call I have FreeSwitch bridge this leg to an outbound leg destined to a customer (via our carrier). The agent listens as this external number is dialed then is prepared to speak with that person. So all is well except a fraction of these calls fail at the step where FreeSwitch calls the agent. There seems to be a 'hangup' event received from the client at the moment this leg is answered. This happens maybe 1/3 of the time. here's my steps 1. call a Lua script via ESL 2. Lua script generates a call leg to agent: local session = freeswitch.Session("user/" .. agentExt .. "@ stuff.com") session:sleep(1000) if session:ready() then session:execute("pre_answer") session:execute("ring_ready") session:execute("sleep", "1000") session:setVariable("outbound_callee_number", outboundCalleeNumber) session:setVariable("outbound_caller_number", outboundCallerNumber) session:setVariable("outbound_caller_name", outboundCallerName) session:transfer("sales_outbound", "XML", "telifi") ******** fails here ~30% of the time - phone rings on the agent's desk, she answers and the call immediately ends ******** 3. this leg is transferred to an XML dialplan that handles bridging to a customer's phone number: <extension name="sales_outbound"> <condition field="destination_number" expression="^sales_outbound$"> <action application="log" data="INFO connected to agent, now dialing out..."/> <action application="sleep" data="1000"/> <action application="playback" data="ivr/8000/ivr-call_being_transferred.wav"/>--> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="effective_caller_id_number=${outbound_caller_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_name}"/> <action application="sleep" data="1000"/> <action application="bridge" data="sofia/gateway/mycarrier/${outbound_callee_number}"/> </condition> </extension> I appreciate any insight or suggestions you could provide.

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Re: [Freeswitch-users] very long delays in conference audio
by Peter Olsson 18 Mar '11

18 Mar '11

Is the delay just in DTMF, or do you know for sure that audio is delayed the same way as well? Do you have jitter buffer enabled - it will delay the audio some. Also, is it always the same delay, or does it build up over time? /Peter ----- Reply message ----- Från: "covici(a)ccs.covici.com" <covici(a)ccs.covici.com> Datum: fre, mar 18, 2011 07:08 Rubrik: [Freeswitch-users] very long delays in conference audio Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> I updated to latest git and still the delay. Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > yes there was a regression on march 2nd that was fixed about the 9th > that could explain your symptoms but I would update and find out for > sure. > > > On Thu, Mar 17, 2011 at 8:32 PM, <covici(a)ccs.covici.com> wrote: > > It seems to be somewhat new, I am on FreeSWITCH Version 1.0.head > > (git-5640464 2011-03-08 12-40-58 -0600) -- I can certainly try to update > > and see if that helps -- I don't like to update weekly, unless its > > necessary. I think I was on something from 2/11 before this one. > > > > > > Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > > > >> What revision of GIT are you on? Is this s new development for you? > >> > >> > >> On Thu, Mar 17, 2011 at 6:49 PM, <covici(a)ccs.covici.com> wrote: > >> > no loopback used here for that. > >> > > >> > George Niculae <george.niculae79(a)gmail.com> wrote: > >> > > >> >> 2011/3/18 <covici(a)ccs.covici.com> > >> >> > >> >> > Yep, real hardware. I wonder if its my itsp, but since it happened on > >> >> > the fs conference, I doubt it. > >> >> > > >> >> > > >> >> I experienced audio delay in fs conferences some time ago when bridging conf > >> >> and using mod_loopback. Is this also your case? > >> >> > >> >> George > >> >> > >> >> ---------------------------------------------------- > >> >> Alternatives: > >> >> > >> >> ---------------------------------------------------- > >> > > >> > -- > >> > Your life is like a penny. You're going to lose it. The question is: > >> > How do > >> > you spend it? > >> > > >> > John Covici > >> > covici(a)ccs.covici.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 > >> > > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale@hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888@conference.freeswitch.org > >> googletalk:conf+888@conference.freeswitch.org > >> pstn:+19193869900 > >> > >> _______________________________________________ > >> 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 > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > covici(a)ccs.covici.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 > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.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 !DSPAM:4d82f64d32765360412282!

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Questions of packages loss
by joy this 18 Mar '11

18 Mar '11

I have some tests for the problem that the voice keeps off and on during the call. I use Wireshark to check the package flow on the caller, callee, and FS sides. The network structure: caller(PCMA)---(Wi-Fi)--->WAN--->FS--->callee(PCMA). FS and the callee are under the same router. The result is: The caller sends 2688 packages to FS, and FS receives 2687 packages. Then FS sends only 2380 packages to the callee, and the callee receives 2380 packages. There is about 11.4% package loss in FS, and the callee hears incomplete voice. I have some questions: 1. I guess FS drops packages from the caller because these packages arrived too late. Is it reasonable? If not, what is the reason? 2. I want to check if FS drops packages because of the late arrival. Could you tell me the packages flow in the source code? 3. I set the jitterbuffer_msec from 10 to 10000, but it seems not effective. Wouldn’t the jitterbuffer_msec affect the result? Sincerely yours, thisjoy.

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very long delays in conference audio
by covici＠ccs.covici.com 18 Mar '11

18 Mar '11

Hi. I have been experiencing very long delays in conference audio -- usually over the internet. For instance, it can take 1 second or more between the time you hit 0 and when it says you are muted or unmuted. I even had this problem during the freeswitch conference. What can be done about this -- any way to tell why this is happening, because it makes conversation on the calls very difficult. I had one last night where they gave up because of this problem. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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mod_conference question
by DJB International 18 Mar '11

18 Mar '11

I wonder whether there is a way to get Conference-Unique-ID to show up when the channel is hang up in CS_REPORTING state. I am trying to find a way to separate and group the cdrs for each conference session. Thank you, -djbinter

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