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having problem getting return value from lua script
by Vincent Xia 10 Jun '13

10 Jun '13

when i was trying to get return value from lua script (with which sql query was processed), <action application="set" data="myvar=${expand(luarun GetDataFromDB.lua ${caller_id_number})}"/> <action application="info" data="myvar=${myvar}"/> GetDataFromDB.lua -------------------------------------- callerid = argv[1] -- DB sql query ... varout = callerid..'abc' return varout the info shows "myvar=+OK.......", then i tried <action application="set" data="myvar=${expand(lua GetDataFromDB.lua ${caller_id_number})}"/> -- using lua GetDataFromDB.lua instead of luarun GetDataFromDB.lua the info shows an empty string, "myvar=". how to get the return value from lua script?

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how to set up calling number in freeswitch.Session
by Vincent Xia 10 Jun '13

10 Jun '13

hi all, while initiating a call from lua with session = freeswitch.Session("user/"..dest); the calling number shown by the called party is "000000000" is it possible to set the calling number to the actual number?

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mod_distributor
by Sayyed Mohammad Emami Razavi 09 Jun '13

09 Jun '13

after installing mod_distributor freeswitch cli gave me: freeswitch@internal> distributor_ctl reload -ERR distributor_ctl reload Command not found!

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Most useful modules (was: SIP web client recommendations)
by Matthew Cordes 09 Jun '13

09 Jun '13

Ken, Thanks for the recommendation. I tried mod_rtmp and it works pretty well my needs. Are there any other modules that are of particular interest to someone new to Freeswitch? I've seen the praises for mod_curl_xml and I'm integrating that into my project. Are there any other modules I should keep on my radar? -Matt ---------- Forwarded message ---------- > From: Ken Rice <krice(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 08 Jun 2013 18:28:21 -0500 > Subject: Re: [Freeswitch-users] SIP web client recommendations > You could look at mod_rtmp > > > On 6/8/13 6:17 PM, "Matthew Cordes" <cordes.matthew(a)gmail.com> wrote: > > Hello, > > Could anyone recommend a SIP web client (either using flash or Java > applets) that they found worked particularly well with FreeSwitch. I've > tried a few of them with mixed results. http://phono.com is the best one > I've tried, but one of my requirements is that the client (and media server > if needs one like flash) be deployable inside my company's intranet w/o > access to the internet. This rules out all of the cloud based solutions > sadly. > > I'm also playing with WebRTC but I don't think that's mature enough for my > immediate needs. > > I'm not biased against a commercial solution. > > -Matt <http://www.freeswitch.org> > > >

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fs no longer works with mono 3.0.7
by covici＠ccs.covici.com 09 Jun '13

09 Jun '13

I am using fs at 3e4fb4b0a265e727e65a826e161f51d4f404f43a of March 6, and it no longer works with mono 3.0.7 which is now my default using gentoo. Will updating to git fix, or should I file a jira? The first message I get is: 2013-06-09_12:01:44.32719 * Assertion at /var/tmp/portage/dev-lang/mono-3.0.7/work/mono-3.0.7/mono/mini/mini-amd64.c:487, condition `amd64_is_imm32 (disp)' not met Thanks. -- 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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INCOMPATIBLE_DESTINATION
by Sayyed Mohammad Emami Razavi 09 Jun '13

09 Jun '13

Now we have some routers and i got INCOMPATIBLE_DESTINATION from AS5350 cisco router after routeing call to the trunk. PLZ give me your guesses about this response and problem. Your help make me free from some big problems in my work. thanks alot.

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 84, Issue 66
by Matthew Cordes 09 Jun '13

09 Jun '13

Thanks Ken, that's what I expected, but thought I'd check anyway. -Matt > ---------- Forwarded message ---------- > From: Ken Rice <krice(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 08 Jun 2013 18:27:57 -0500 > Subject: Re: [Freeswitch-users] Echo cancellation > Theres not really anything you can do from the Switch for this... > Acoustic Echo is always a problem when you are using the built in laptop > mic and speakers unless you have a client that will filter it out > > > On 6/8/13 6:10 PM, "Matthew Cordes" <cordes.matthew(a)gmail.com> wrote: > > Hello, > > What is the best strategy for dealing with echos when a user is placing a > call from a computer's softphone without a headset? In my tests thus far > with a vanilla 1.2.10.x build of FreeSwitch the echo is unbearable when > just using my laptop's microphone. From what I've read echo cancellation is > best done on the client (and perhaps I'm just not finding clients that > support this). Is there a FreeSwitch setting or plugin that can have remedy > the situation? > > -Matt > >

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Echo cancellation
by Matthew Cordes 08 Jun '13

08 Jun '13

Hello, What is the best strategy for dealing with echos when a user is placing a call from a computer's softphone without a headset? In my tests thus far with a vanilla 1.2.10.x build of FreeSwitch the echo is unbearable when just using my laptop's microphone. From what I've read echo cancellation is best done on the client (and perhaps I'm just not finding clients that support this). Is there a FreeSwitch setting or plugin that can have remedy the situation? -Matt

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Using mod_xml_curl from parse.com
by Ivan C Myrvold 08 Jun '13

08 Jun '13

I am thinking about using parse.com to serve dynamic configuration XML files for FreeSWITCH. parse.com gives this as an example of how to use Cloud Code to run a function: curl -X POST \ -H "X-Parse-Application-Id: wCjhhG6tSASMGxvFwPESUCFCW3KykGeyCJqg8z1U" \ -H "X-Parse-REST-API-Key: wL8WMeyLcKyfemomB94mLICLhhi8xw1G1DrEapAp" \ -H "Content-Type: application/json" \ -d '{}' \ https://api.parse.com/1/functions/hello Is it possible to populate xml_curl.conf.xml so that freeswitch will use the same HTTP headers? Ivan

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Call Forwarding Name Display issue
by Sherwood Hu 08 Jun '13

08 Jun '13

I have two Cisco SPA504G phones that are connected to freeswitch. I am still in the testing stage. Today I tested call transfer scenario: Incoming call -> IVR -> extension 1 -> transfer -> extension 2 I used the 'transfer' button on the phone to transfer the call. On the second phone, it displayed the name of extension 1, not the caller number. What I wanted is that the caller number displayed on the phone 2. What's wrong here?

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