- FreeSWITCH-users - FreeSWITCH Mailing Lists

help
by Joshua Shaffner 21 Mar '11

21 Mar '11

On Mon, Mar 21, 2011 at 4:10 PM, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Mod_fifo outbound strategy enterprise (Marc de Corny) > 2. Does Freeswitch complies with SIP Connect technical > recommendation? (Yehavi Bourvine) > 3. unable to insert channel variable in new call leg w/in Lua > script (elijah) > 4. Re: New FreeSWITCH Module: mod_ladspa (Frank Carmickle) > 5. Delayed start of tone_detect (Stephen Wilde) > 6. Re: New FreeSWITCH Module: mod_ladspa (Michael Collins) > 7. Re: Understanding break="never" in condition-tag (Michael Collins) > > > ---------- Forwarded message ---------- > From: Marc de Corny <marcdecorny(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 17:59:16 +0000 > Subject: [Freeswitch-users] Mod_fifo outbound strategy enterprise > Hi all, > I am getting the hang of the fifo now and it is proving very stable which is what we all need. > Got one question however regarding mod_fifo and the outbound strategies. > From looking at the outbput of the fifo list commands I can see that that parameter is always set to "ringall". which is fine, but I can also see many stats on that number of calls and the last call that the members have taken. Is there also a way of sending the calls to the longest idle or something of that nature? Is that done by setting the outbound_strategy to "enterprise" or is there another way to achieve that. > Is there also a command to change that value without editing the XML as I am trying to make everything dynamic. > any input is very much appreciated. > thanks > Marc > > ---------- Forwarded message ---------- > From: Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 20:46:20 +0200 > Subject: [Freeswitch-users] Does Freeswitch complies with SIP Connect technical recommendation? > Hello, > > We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present. > > Does Freeswitch fullfills all the recomendations there? > > Thanks, __Yehavi: > > ---------- Forwarded message ---------- > From: elijah <elijah(a)crankenstein.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 11:50:44 -0700 > Subject: [Freeswitch-users] unable to insert channel variable in new call leg w/in Lua script > In a Lua script I instantiate a new call leg and (attempt to) insert a new channel variable: > local session = freeswitch.Session("user/1001(a)stuff.com") > if session:hangupCause() == "SUCCESS" then > session:execute("export", "my_variable=" .. myVariable) > session:transfer("ext_in_mycontext", "XML", "mycontext") > However, I am not able to retrieve my custom channel variable later. This call leg is established correctly but there is no point in this call leg's life where I can seem to find my variable. Is there another method by which I should try to insert a channel variable in a newly created call leg? > > > > ---------- Forwarded message ---------- > From: Frank Carmickle <frank(a)carmickle.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 15:27:39 -0400 > Subject: Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa > Hey Michael > This is cool. Has there been any thought to making freeswitch a jack client. That IMHO would be very very useful. That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound. > Thanks > --FC > On Mar 18, 2011, at 6:56 PM, Michael Collins wrote: > > Hello all! > The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story: > http://www.freeswitch.org/node/313 > Have fun! > -Michael > _______________________________________________ > 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 > > > > ---------- Forwarded message ---------- > From: Stephen Wilde <wstephen80(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 20:33:28 +0100 > Subject: [Freeswitch-users] Delayed start of tone_detect > I start the tone_detect on answer using "execute_on_answer" as: > <action application="export" data="nolocal:execute_on_answer=tone_detect MYTONE 680 ro +5000 set MYTONE=1" /> > but I have the necessity to start the tone_detect for example 500ms after answer: there is a way to do that in dialplan? > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 12:39:18 -0700 > Subject: Re: [Freeswitch-users] New FreeSWITCH Module: mod_ladspa > Not that I know of, but "patches welcome"!! :) > -MC > > On Mon, Mar 21, 2011 at 12:27 PM, Frank Carmickle <frank(a)carmickle.com> wrote: >> >> Hey Michael >> This is cool. Has there been any thought to making freeswitch a jack client. That IMHO would be very very useful. That would allow sip, or any other protocol that fs supports, to be a way to get audio in to professional sound production software like ardour, audacity or ecasound. >> Thanks >> --FC >> On Mar 18, 2011, at 6:56 PM, Michael Collins wrote: >> >> Hello all! >> The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story: >> http://www.freeswitch.org/node/313 >> Have fun! >> -Michael >> _______________________________________________ >> 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 >> >> >> _______________________________________________ >> 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 >> > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 21 Mar 2011 13:10:26 -0700 > Subject: Re: [Freeswitch-users] Understanding break="never" in condition-tag > >> >> I can't/didn't contribute much to this discussion, but I do follow it. My idea and (little) testing experience supports the explanation of Dimitry. >> Either way, it 'proofs' there is confusion about the working of the 'break' in condition tags. >> If I can find the time, I will try to do some tests / build an example myself. Thank you both for the answers and insights so far. >> > > Okay, for those of you who have the bridge book, please go to page 167-168. Darren did a really good job of explaining this. For those of you who don't have the book, SHAME! :) > The break flag determines what happens on processing conditions within a single extension. Let's say that chan var ${freeswitch} has the value "rocks" and ${foo} has the value "bar". > <extension name="break_flag_is_not_confusing"> > <condition "${freeswitch}" expression="^lame$"> >  > </condition> >  > <condition "${foo}" expression="bar"> > ... > </condition> > </extension> > Look at the above snipped. The first condition evaluates to false, which means all processing for this particular extension stops. The parse "breaks" out of this extension and moves on to the next one. Now look at this extension with the the break-flag set to "never"... > <extension name="break_flag_is_not_confusing"> > <condition "${freeswitch}" expression="^lame$" break="never"> >  > </condition> >  > <condition "${foo}" expression="bar"> > ... > </condition> > </extension> > See the difference? With the break flag set to "never" then it does not matter whether the condition is evaluated as true or false - the extension's next condition gets evaluated. Why do you need the break="never"? Simple: when no "break" is specified there is an implied break="on-false" for the condition. In other words, by default, if one condition inside of an extension evaluates to false then all processing for that extension stops. This has the effect of allowing you to "stack" conditions to create a logical AND. > So, in summary, the break flag will control how the parser behaves when evaluating conditions. Normally when a single condition fails, the whole extension is skipped, however when you do break="never" on a condition then it does not matter if that particular condition is true or false - extension processing will continue. > Hope this helps. If it doesn't then read chapters 5 and 8 of the book. Chapter 5 (me) is a gentle intro into the dialplan. Chapter 8 (Darren) talks more extensively about the advanced concepts of dialplan processing. > -MC > > _______________________________________________ > 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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pass variable from dialplan to bridge
by dunchan 21 Mar '11

21 Mar '11

Hi! I have a simple question, how can i pass a varible from dialplan to the bridge? I want to fill the 'from-user' field depends on sip user settings. I've tryed the following in dialplan: ... some conditions ... <action application="set" data="from-user=${outbound_caller_id_number}"/> <action application="bridge" data="sofia/gateway/upc/$1"/> in a gateway xml i tryed: <gateway name="upc"> <param name="username" value="..."/> <param name="password" value="..."/> <param name="proxy" value="..."/> <param name="register" value="false"/> <param name="from-user" value="${from-user}"/> </gateway> Above config ha no effect :( any suggestions? thx, Viktor

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Recommendations for multi-channel provider of Singapore DID's
by Chris Fowler 21 Mar '11

21 Mar '11

Hi, We use InPhonex for International DID's. Unfortunately in Singapore they are unable to provide DID's with more than 2 Channels. Can anyone recommend a provider that can provide DID's with more than two channels in Singapore (we need 4 - 20 Channel DID's)? Thanks, Chris.

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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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