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Re: [Freeswitch-users] dialing multiple destinations simultaneously in lua
by Kalyani Kulkarni 19 May '14

19 May '14

Hi Vincent, Create outbound ESL connection with async property as per wiki <action application="socket" data="127.0.0.1:8084 async full"/> Once the call is routed to lua socket server that you run on specified port you can execute almost everything con:execute("application", "params"); For your case, application will be bridge and params will be user/1001, user/1002 hope this helps. Kalyani ---------- Forwarded message ---------- > From: Vincent Xia <gmangudai(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Fri, 16 May 2014 18:22:47 +0800 > Subject: [Freeswitch-users] dialing multiple destinations simultaneously > in lua > here is what the wiki says about how to dial multiple destinations > simultaneously: > > “Multiple destinations in outbound calls. This means any dialstring > may contain a '|' and/or ',' separated list of call URLs. The "|" > calls one call at a time, the "," calls multiple calls simultaneously. > The example below will call 2 locations and connects the first to answer > > <extension name="3002"> > <condition field="destination_number" expression="^3002$"> > <action application="bridge" > data="iax/guest@somebox/1234,sofia/test-int/1000@somehost"/> > </condition> > </extension>” > > so my question is what's the equivalent of doing something like > > <action application="bridge" data="user/1001, user/1002"/> > > in lua script? though i wonder if it's possible cuz someone has said > the scripts are not asynchronous, they work in > > blocked mode so the calls will have to be made one after another, not > simultaneously. > > >

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how to capture variables from lua script in xml dialplan?
by Vincent Xia 17 May '14

17 May '14

the dialpan is as follow: <extension name="Test_Extension"> <condition field="destination_number" expression="^\b(10[0-1][0-9])\b$" break="never"> <action application="set" data="continue_on_fail=true" /> <action application="lua" data="setvar.lua" /> <action application="log" data="INFO my_lua_var=${my_lua_var}" /> </condition> <condition field="${my_lua_var}" expression="1"> <action application="sleep" data="1000" /> </condition> </extension> the the xml dialplan i have a variable set inside a lua script(session:setVariable("my_lua_var", "1");), but its value is always "undefined" during the parsing stage. i need the variable to be a condition field so how can i capture it?

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Proxy Refer
by Gareth Rylance 17 May '14

17 May '14

I have the following A core PBX that controls call flows and selects an end target. The core PBX hairpins the calls through a Freeswitch instance that for example records the calls. The targets use REFER messages to transfer calls between each other & third parties. Is it possible to configure the service freeswitch instance to proxy the REFER messages back to the core PBX and not convert them into an INVITE? So that the message flow looks like the attached picture. The Refer message needs to be proxied in order to tear down the hairpin as the calls are transferred. Thanks Gareth [image: Inline images 1] SD Edit source for image http://sdedit.sourceforge.net/index.html ----------- alice:Actor "Alice" core:core[rp] "core" service:service[rp] "service" t1:t1[rp] "target 1" t2:t2[rp] "target 2" alice:core.Invite(12345567) core:service.Invite(12345567) service:core.Invite(12345567) core:t1.Invite(12345567) t1:core.200 ok core:service.200 ok service:core.200 ok core:alice.200 ok t1:core.Refer(56789) core:service.Refer(56789) service:core.Refer(56789) core:service.Invite(56789) service:core.Invite(56789) core:t2.Invite(56789) --------------

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Can I disable rport?
by Richard Brady 17 May '14

17 May '14

Hi folks Can I disable rport on outbound TCP SIP requests? The rport Via parameter is of course a good thing to set on outbound SIP requests when using UDP but I think it's a bad thing when using TCP. This is because outbound TCP requests use a dynamic source port and the UAS will not be able to establish a new TCP connection to this port if it needs to, for example to send a response after the connection has been closed. In most cases this won't be needed - hopefully the original connections remains open - but with mobility, firewalls, etc, connections do sometimes get closed between initial request and final response. Regards, Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

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att_xfer to call group
by Karl Schmidt 17 May '14

17 May '14

Is this possible? att_transfer wants a channel_url for an argument - I want to give it a call group? att_xfer <channel_url> -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 All governments border on being worse than no government. It is when the people start thinking that their government can protect them from some evil that government crosses the line and becomes worse than no government. -kps --------------------------------------------------------------------------------

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Nortel CS1000 to FreeSWITCH
by Joel White 16 May '14

16 May '14

I am still working on integrating Nortel with FreeSWITCH I have the calls coming in through FreeSWITCH and parsing/matching XML Here is the latest issue though the call runs through XML, looks like it is working then the calls drops and no audio was ever heard on the Nortel extension calling in I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH The call hangs up with an reason 79 https://wiki.freeswitch.org/wiki/Hangup_Causes I checked and the audio is PCMU:8000 What am I missing here? Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH Thank you in advance for any assistance Joel

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A few questions regarding mod_skypopen
by ThunderZ 16 May '14

16 May '14

Hi, there I have a few questions regarding this mod_skypopen module. 1. While loading the module, if one skype interface fails to load for whatever reason, i.e. the skype user didn't login for that interface or Xvfb server had a issue, why all the other working interfaces are removed automatically, and the module itself is unloaded with an error? I don't think a single dead interface should affect other working ones. 2. According to the Wiki, "sk reload => this re-reads Skypopen configuration file skypopen.conf.xml and adds ONLY new interfaces it found in conf." However, when I do a "sk reload", I found all the new interfaces are ignored too. Did I miss something? I'm using the latest git version. 3. With the default skype configuration template, I found that some skype accounts always return this "/P2P connect failed" error using the standard startup script. However, if I use the same command to login such accounts from a real X server, it will work ok (abeit after some considerable delay). After that, it also works ok with the Xvfb display via the standard script. / Any advice will be greatly appreciated./ Thanks. /

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more skypopen questions
by Russell Treleaven 16 May '14

16 May '14

Hi List, I have a few more questions about skypopen. q: why are the symlinks created for the instances? a: q: why are the passwords not stored in the xml skypopen.conf.xml a: q: How do I start skypopen.ko module from modbrope.d? a: Sincerely, Russell Treleaven rtreleaven(a)bunnykick.ca

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OpenBSD/NetBSD Updates
by Brian West 16 May '14

16 May '14

I’ve updated the build instructions for both OpenBSD and NetBSD: https://confluence.freeswitch.org/display/FREESWITCH/OpenBSD https://confluence.freeswitch.org/display/FREESWITCH/NetBSD Please feel free to improve the docs for these, Seems in OpenBSD 5.5 their exists no OpenSSL that will work with FreeSWITCH so we have to build our own for now. -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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problem with supported header
by Jason Holden 16 May '14

16 May '14

Currently as for RFC precondition should not be included in the support header unless 100rel is enabled. As you can see below precondition is in the list even though 100rel is not enabled. Is this corrected as of a specific git rev for the stable 1.2 branch? Supported: timer, precondition, path, replaces.

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