January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using Freeswitch with TLS and without TLS together
by Yücel ALTUNAY 05 Feb '15

05 Feb '15

Hi, I want to use freeswitch with TLS on my mobile phones and without TLS on my GSM gateways together. I see some examples but i coudn't do it. Is someone has any example to do this? Thank you.

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freeswitch+ovz - poor sound, terrible timings
by Денис 05 Feb '15

05 Feb '15

timer_test 20 200 ... 2015-01-29 15:26:32.717637 [CONSOLE] mod_commands.c:846 Timer Test: 96 sleep 20 19975 2015-01-29 15:26:32.738371 [CONSOLE] mod_commands.c:846 Timer Test: 97 sleep 20 35287 2015-01-29 15:26:32.738371 [CONSOLE] mod_commands.c:846 Timer Test: 98 sleep 20 4676 2015-01-29 15:26:32.777674 [CONSOLE] mod_commands.c:846 Timer Test: 99 sleep 20 20051 timer_test 10 200 ... 2015-01-30 20:17:50.717723 [CONSOLE] mod_commands.c:846 Timer Test: 184 sleep 10 10084 2015-01-30 20:17:50.754812 [CONSOLE] mod_commands.c:846 Timer Test: 185 sleep 10 37157 2015-01-30 20:17:50.760749 [CONSOLE] mod_commands.c:846 Timer Test: 186 sleep 10 6206 2015-01-30 20:17:50.760749 [CONSOLE] mod_commands.c:846 Timer Test: 187 sleep 10 9680 2015-01-30 20:17:50.777621 [CONSOLE] mod_commands.c:846 Timer Test: 188 sleep 10 6890 2015-01-30 20:17:50.777621 [CONSOLE] mod_commands.c:846 Timer Test: 189 sleep 10 10005 On host node - all ok. Timers - with posix, soft, timerfd... What can i do?

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can I use Event_Socket_Outbound to only monitor?
by Shai Perelman 05 Feb '15

05 Feb '15

hi, I am using https://wiki.freeswitch.org/wiki/Event_Socket_Outbound my goal is to monitor what happens on the extension but without affecting the normal behaviour. I have this dialplan. TagTypeDataOrder[image: add] <http://148.251.142.45/app/dialplan/dialplan_detail_edit.php?id2=8a56aba2-5c…> condition username 9998 1[image: edit] <http://148.251.142.45/app/dialplan/dialplan_detail_edit.php?id=02e9a3bf-15c…> [image: delete] <http://148.251.142.45/app/dialplan/dialplan_detail_delete.php?id=02e9a3bf-1…> action socket 109.65.149.22:9999 async 3[image: edit] <http://148.251.142.45/app/dialplan/dialplan_detail_edit.php?id=0ca4a9d4-020…> [image: delete] <http://148.251.142.45/app/dialplan/dialplan_detail_delete.php?id=0ca4a9d4-0…> I am calling *9664 (music on hold) and the call is blocking while wating for commands from my side. I want it to answer automatticaly like it would do with out the socket module, and just send me the events regarding that extension thanks Shai ITD Communications

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Need help with Video proxy-media mode
by Cinthia Leung 05 Feb '15

05 Feb '15

Hi there, I'm trying to setup FS and proxy media. Here's my setup. Kamailio SBC <-> FS <-> MCU. >show codec sees H.264 as passthru late-negotiation = true Audio is ok but I get no video. Tshark capture shows that video streams enter FS from both Kamailio and MCU. FS was just not forwarding video packets to the other side. Codec is H.264. Tested with FS Version 1.14.15 and 1.5.15b+git~20150130T165344Z~4ed7b4811a~64bit Bypass media works fine. Being able to proxy media is a requirement for us. I hope it's just something obvious that I missed. I'll provide logs if needed. Thanks in advance! Cindy

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Convert existing call to a conference and hangup the old call
by Kumar, Sharath 04 Feb '15

04 Feb '15

Hello, I am new to Freeswitch and wanted some input on how best to achieve the below scenario. 1. A and B are in a call. 2. A adds user C to the call. - At this point user A and B are added to a dynamically created conf and C is added as well. 3 A adds user D to the call. D is added to the conf. The fact that step 2 is a conf call and users C and D are provisioned and available to the FS. I am using FS_CURL so the PHP has access to the DB. I know how to add A,B,C and D directly into a conference using mod_conference. I made it work. However, I don't understand how I can access an existing call i.e between A and B and turn that into a conference. and also I guess the original call should be hungup when B accepts the conf invite. Is there a way to do this in the dial-plan ? i.e when A calls C, the PHP knows that this is a conf call, how do I kill the old call though. I guess I can use the uuid of the previous call, I don't know what to issue to hang up the first call. My clients are webrtc based and currently they don't natively support either hold/resume or transfer(using refer). They do however support multiple calls from/to the same user at the same time. i.e 2 lines. I guess both will be active. Thank you. Any help would be greatly appreciated!! -Shaks.

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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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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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