February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call intercept problem
by Tim Bock 03 Feb '15

03 Feb '15

Hello list, Running freeswitch 1.4.6, though have also seen this issue with older versions. After the initial freeswitch install, I tested the call intercept function, and it worked fine. Fast forward several months ahead, and now call intercept doesn't work. Looking at the logs, the message is that the channel is no longer available. Indeed, when I examine the uuid the call intercept function is trying to connect, the uuid is for the call immediately preceding the current call, sort of like an "off by one" indexing error. This is reliably repeatable. Any thoughts on why this is occurring? And more importantly, any way to fix it? Thank you, Tim -- Tim Bock Synacktics, LLC www.synacktics.com (505)795-1511

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glibc GHOST vulnerability
by Oleg Stolyar 03 Feb '15

03 Feb '15

Sorry if the question is naive - trying to be paranoid here. On my CentOS machines I updated my glibc version to one that fixed the GHOST vulnerability. Do I need to rebuild FS or is the library linked dynamically, so there is no need to rebuild?

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Re: [Freeswitch-users] Issues with originate, webrtc and xml_curl
by Benjamin Rowe 03 Feb '15

03 Feb '15

I've been trying to dig into this more but can't find the issue i think it must have something to do with the dial-string. Here is curl response from my PHP script for 1004's authentication. <?xml version="1.0" encoding="utf-8" ?><document type="freeswitch/xml"> <section name="directory"> <domain name="0.0.0.0"> <groups> <group name="default"> <users> <user id="1004"></user> </users> <params> <param name="dial-string" value="{presence_id=1004@0.0.0.0}${sofia_contact(1004@0.0.0.0)}"></param> <param name="a1-hash" value="<HASH>"></param> </params> <variables> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Test"></variable> <variable name="effective_caller_id_number" value="1004"></variable> <variable name="dtmf-type" value="info"></variable> <variable name="toll_allow" value="domestic,international,local"></variable> <variable name="accountcode" value="1004"></variable> <variable name="transfer_fallback_extension" value="operator"></variable> </variables> </group> </groups> </domain> </section></document> On Thu, Jan 29, 2015 at 6:20 PM, Benjamin Rowe <you.kissmyarse(a)gmail.com> wrote: > Evening All, > > I have a basic system up and running i am able to dial out via my provider > on both sip endpoints. I have setup 2 users 1000 and 1004. 1000 is created > from the flat file xml and 1004 is from the xml_curl request. when i > attempt to originate a call from user 1000 to an external number it passes > through fine however when i originate a call from user 1004 i get > subscriber absent both are running the same setup to my knowledge. Any > Ideas? > > Errors: > 2015-01-29 18:12:17.745051 [WARNING] mod_dptools.c:3979 Can't find user > [1004@<IP>] > 2015-01-29 18:12:17.745051 [NOTICE] switch_ivr_originate.c:2735 Cannot > create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] > 2015-01-29 18:12:17.745051 [DEBUG] switch_ivr_originate.c:3723 Originate > Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] > 2015-01-29 18:12:17.745051 [INFO] mod_dptools.c:3234 Originate Failed. > Cause: SUBSCRIBER_ABSENT > 2015-01-29 18:12:17.745051 [NOTICE] switch_channel.c:4724 Hangup > loopback/1004-b [CS_EXECUTE] [SUBSCRIBER_ABSENT] > > Sofia Registrations: > > ================================================================================================= > Call-ID: ocq57n8fn23uo0s7rh2025 > User: 1004@<IP> > Contact: "" <sip:mp2337gf@1tfoakfdb6me.invalid > ;transport=ws;fs_nat=yes;fs_path=sip%3Amp2337gf%40<IP>%3A59613%3Btransport%3Dws> > Agent: SIP.js/0.6.3-devel BB > Status: Registered(WS-NAT)(unknown) EXP(2015-01-29 18:24:02) > EXPSECS(571) > Ping-Status: Reachable > Host: fs01-a > IP: <IP> > Port: 59613 > Auth-User: 1004 > Auth-Realm: <IP> > MWI-Account: 1004@<IP> > > > Originate command: > originate > {ignore_early_media=true,origination_caller_id_number=1004}loopback/1004 > &bridge(sofia/gateway/test/555123456) > > Many Thanks > Benjamin Rowe > -- Benjamin Rowe Lazypeople Mobile: +44 (0)7904 026869 Phone: +44 (0)845 86 99 892 Email: ben(a)lazypeople.co.uk Web: http://www.lazypeople.co.uk

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Help need for Lua Script development
by Mayank Nakrani 03 Feb '15

03 Feb '15

Hello Guys, Im starting with my Lua scripts dev. Have already had luck writing an Callback script. Now i would to go deeper integrating it with Mysql and other libraries and generating some dynamic dialplan. Till now i used to write my scripts in Text editor and would to know is it Possible in IDE like Eclipse. I installed Lua development plugin in Eclipse but was wondering Whats the correct way to import freeswitch libraries in the project so that i can see methods/functions in in freeswitch API for easier and faster coding. I tried loading "libfreeswitch" and "mod_lua" lib using "require" command but its failing with [com.naef.jnlua.LuaRuntimeException: error loading module 'libfreeswitch' from file './libfreeswitch.so': ./libfreeswitch.so: undefined symbol: apr_os_default_encoding ] Im gud in Java and can use mod_java but would like to stick with lua as i heard its more closer to freeswitch. Any help :) Regards Mayank Nakrani

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fs_cli poll or select
by akhil garg 03 Feb '15

03 Feb '15

In freeswitch fs_cli is using poll() when linux and for windows it is using select(). what is the specific reason? How can I use select()? -- regards, akhil

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How to force park timeout for not bridged calls?
by Mateusz Bartczak 03 Feb '15

03 Feb '15

Hi I have problem with park application and forcing FS to use timeout I tried two configurations: <action application="set" data="park_timeout=30"/> <action application="park"/> And second: <action application="set" data="api_result=${sched_api(+30 park_timeout uuid_kill ${uuid} CRASH)}"/> <action application="park"/> None of them works. After 30 seconds of having call in park state nothing happens, I see no messages on console related to sched_api or park_timeout execution Maybe timeouts work only on bridged calls? My intenion is to have timeout for calls that I put in park and they have been not handled properly (ie bridged) by external script for some time. Such calls should end automatically after given time

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FreeSWITCH Week in Review (Master Branch) January 24th-31st
by Ken Rice 02 Feb '15

02 Feb '15

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1tX1HQN FreeSWITCH Week in Review (Master Branch) January 24th-31st Hello, again. This week in the FreeSWITCH master branch we had 17 commits. The features this week are: an additional tweak to avoid jumping back and forth on ICE and setting ANI plan and ANI type for ftmod_libpri . New features that were added: 8d599a8 One more tweak to not jump back and forth on ice when you have 2 reachable 8307e78 Set ANI plan and ANI type for ftmod_libpri 3cd62df If ANI ton is not interpreted correctly by libpri, fallback to calling ton/type. The following bugs were squashed: FS-7193 Fix for sofia contact being encoded which makes impossible to call a registered user FS-7191 Edit pgsql example connection string to remove unneeded option that may cause a failure on some systems b87d42c Fix read of 0 bytes problem and make buffer null terminated before using strstr FS-7205 Do not url encode unless an at sign is in the uri FS-7211 Fix for sofia_contact returns unable to locate registered user in latest master build FS-7208 _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp FS-7214 Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache c8f8d6b Fix a null pointer dereference in the native postgresql connection code. This is unlikely to ever be hit in the field. FS-7217 Use upper case when you query e062f5e If the span has been already fully stopped and ftdm is not running, return success from the span stop function.

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SRTP with AES 256
by Melanie Treitinger 02 Feb '15

02 Feb '15

Hi, I'm trying to make sip calls with encryption. Everything works fine with AES 128, but with AES 256 there is no audio but also no error reported. How do I use AES 256? Is it implemented in FreeSWITCH already? Thanks Melanie

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Correct progress time variable in CDR
by John Nash 01 Feb '15

01 Feb '15

In my setup when Invite comes to freeswitch dialplan plays a sound file and then Bridge the call to destination. I am using csv CDR and trying to get timestamp when there is either 180 or 183 from end destination (After bridge). What variables should I use the correct progress timestamp. I printed variables in two extensions .. Extension 1 (where sound file is being played) ------------------------------------------------------------------ Caller-Profile-Created-Time: [1422696316256090] Caller-Channel-Created-Time: [1422696316256090] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Extension 2 (where call is being bridged transferred from Extension 1) ---------------------------------------------------------------------------------------------------- Caller-Profile-Created-Time: [1422696320776083] Caller-Channel-Created-Time: [1422696316256090] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1422696316256090] >From this log I think progress timestamp is the one generated when sound file plays but I need the progress time when 180/183 received after bridge.

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