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FreeSWITCH Week in Review (Master Branch) February 7th-13th
by Ken Rice 16 Feb '15

16 Feb '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1DZDdce FreeSWITCH Week in Review (Master Branch) February 7th-13th Hello, again. This past week in the FreeSWITCH master branch we had 24 commits. The new features are: the addition of mongo_find_n API and the addition of error logs in mod_java. This edition of the FreeSWITCH Week in Review marks the one year anniversary since I started this project! A whole year worth of commits! That is 1701 commits to be exact! And so many more to come! Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7265 Add mongo_find_n API FS-7269 Add error logs in mod_java. Improvements to the documentation: FS-7270 Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile The following bugs were squashed: FS-7236 Fix mutex use before init error caused by 27c8622 0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed. FS-7256 Fix for being unable to load mod_java. FS-7252 Fix for 6-year-old regression from commit 525f1ac back in 2008 FS-7260 Fix for L16@16000h with Asterisk negotiation issue. FS-7264 Fix signed/unsigned warnings on windows building ws.c FS-7236 Re-factor to fix audio problem from commit 7c63670

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csta , anyone experience with it?
by davy van de moere 16 Feb '15

16 Feb '15

I've recently gotten the question how easy it would be to have some support for CSTA on FreeSwitch. I noticed there used to be a module for it a few years ago, and I do see things in mod_sofia.c on csta. What is the good approach on this? Is it doable to based on the event socket to push out/in csta messages? Or do I miss some great resource on the topic? Kind regards, Davy Van De Moere

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B leg is still ringing After FXO caller hangup before ANSWERED
by Charles Wang 16 Feb '15

16 Feb '15

Hi all, I have a server with freeswitch 1.4.15 + freetdm(FXS/FXO). I think there is a bug in inbound call via the freetdm FXO device. When I try to make call from FXO and it bridges to SIP device named 1234 via the following dialplan. <extension name="my call test"> <condition field="destination_number" expression="^(1234)$"> <action application="set" data="call_timeout=30"/> <action application="bridge" data="user/1234"/> </condition> </extension> Before SIP 1234 answers the call, the caller (FXO) hangup call before 1234 answered. But the callee (SIP 1234 or FXS) is still ringing and stop ring until call-timeout about 30 seconds. It is the same condition if the callee is FXS device. I can find two channels during ringing (A leg & B leg). After the caller(FXO) onhook, the A leg is still alive and A leg will not be hangup before the callee (FXS or SIP) stop ring ( call-timeout ). There are two legs (A leg & B leg) after the caller FXO had hangup. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context 53ce4fe9-9511-4625-87ae-1448421c9810,inbound,2015-02-12 17:33:11,1423733591,FreeTDM/2:2/1234,CS_EXECUTE,unknown,unknown,,1234,bridge,freetdm/FXS1/1,XML,TEST,PCMU,8000,64000,PCMU,8000,64000,,charles,,,RINGING,,,,,,,unknown,unknown,,1234,XML,default cf5a7ee4-cbfd-48e4-ab4e-4d757216712c,outbound,2015-02-12 17:33:12,1423733592,FreeTDM/1:1/,CS_CONSUME_MEDIA,unknown,unknown,,1,,,XML,default,,,,,,,,charles,,,RINGING,Outbound Call,1,,53ce4fe9-9511-4625-87ae-1448421c9810,,,unknown,unknown,,1,XML,default Can anyone help me to solve it or tell me why? -- Best Regards Charles

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Bug Marshalls
by Brian West 16 Feb '15

16 Feb '15

We are still in need of community members to help triage, test, confirm issues as they are reported on JIRA, In addition if you see someone posting a bug report to the mailing list, kindly direct them to JIRA to file a bug. If you're interested please contact me or krice(a)freeswitch.org and we can get you setup to assist. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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lua script unable to stop moh
by CT Radu 16 Feb '15

16 Feb '15

Hi all, I have the following problem: I have an dialplan rule in freeswitch that delegates the control for the incoming call on a specific number to an erlang application. This erlang application, at one point executes a lua script using the call: {api, luarun, "luascript.lua uuid"} In lua I do the following: api = freeswitch.API(); session = freeswitch.session(argv[1]); uuid = session:get_uuid(); if session:ready() then uuid_break_cmd = "uuid_break ".. uuid .. " all"; uuid_hold_cmd = "uuid_hold off" .. uuid; freeswitch.consoleLog("notice","uuid break cmd: ".. uuid_break_cmd .. "\n"); session:execute("break","all"); api:executeString(uuid_break_cmd); reply = api:executeString(uuid_break_cmd); freeswitch.consoleLog("notice", "After uuid_break.\n"); end The thing is that freeswitch ignores completly uuid_break and uuid_hold off The same command issued via fs_cli has effect. In fs_cli on debug I can see this: 2015-02-10 20:23:51.405153 [NOTICE] switch_cpp.cpp:1328 uuid break cmd: uuid_break f36d5e5c-b151-11e4-9674-8f19f7458942 all EXECUTE sofia/ctradu-dev.ezuce.ro/1002(a)ctradu-dev.ezuce.ro break(all) 2015-02-10 20:23:51.405153 [DEBUG] mod_erlang_event.c:169 Ignoring event CHANNEL_EXECUTE for attached session f36d5e5c-b151-11e4-9674-8f19f7458942 2015-02-10 20:23:51.405153 [DEBUG] mod_erlang_event.c:169 Ignoring event CHANNEL_EXECUTE_COMPLETE for attached session f36d5e5c-b151-11e4-9674-8f19f7458942 2015-02-10 20:23:51.405153 [NOTICE] switch_cpp.cpp:1328 After uuid_break. Does anyone know why uuid_break has no effect executed from a script launched via api luarun ? (And the same command when issued via fs_cli has effect) The freeswitch version is 1.4.15~64bit Thank you, Costin

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media bypass - freeswitch
by Achintha 16 Feb '15

16 Feb '15

hi I configured freeswitch (FreeSWITCH Version 1.5.15b+git~20150129T212010Z~c8f8d6b964~64bit (git c8f8d6b 2015-01-29 21:20:10Z 64bit)) on centos. I’m using xml_curl module to generate the dialplan. It works properly when bypass_media is set to false. Then I set the media bypass parameter on freeswitch dialplan and then the freeswitch started crashing. I ‘m using attendant transfer feature. Here I have upload core dump file when it crashed to following location. http://pastebin.com/QmTZ665P http://pastebin.com/gsUKzbYu Kindly advise me to solve this problem. -- Best Regards.. Achintha Udukumbura

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mod_managed question
by Malay Thakershi 16 Feb '15

16 Feb '15

Hello all, Deciding whether to go for 32 or 64 bit freeswitch install. I have x86 libraries which are invoked by mod_managed. So if I have x64 mod_managed, can it work with 32-bit libraries? Thanks.

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Freeswitch deployment in SS7 + Sangoma
by Vishal Kakkar 14 Feb '15

14 Feb '15

Hi all, We have been using Freeswitch+Sangoma for PRI deployments(ISDN). But now when we are planning to deploy Freeswitch+Sangoma in SS7 environment, we discovered that there is hefty NSG license cost involved from Sangoma (apart from E1 HW cost). Can anyone please help.. Is there any other possible way to deploy Freeswitch+Sangoma in SS7 environment. Also, looks like Astrisk has libss7 library to make Sangoma Cards work in SS7 without NSG license. Please advice. Thanks a lot, Vishal.

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 13 Feb '15

13 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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RTP delay with cng
by Jose Fco. Irles Durá 13 Feb '15

13 Feb '15

Hi, I have a FreeSWITCH 1.4.15 (from debian repositories), but I've been having a problem for a long time (with older versions too) and I don't know if it is a configuration problem or a bug. The FreeSWITCH runs on an openvz machine and this server is not overloaded. Before testing with trunk branch I want to share my problem looking for help to solve it (it's in production) I have an issue with some providers that send to my FreeSWITCH CN packets. When this happens, the other call leg has a terrible delay and bad quality sound.It starts with a little delay, but during the call the delay increases more and more. Capturing with wireshark and decoding rtp stream I can see 31% of packets dropped by Jitter Buff in the leg in wich the delay occurs (from FreeSWITCH IP to the other endpoint) I don't have any problem when the provider doesn't send CN packets. If I configure FreeSWITCH not to negotiate "comfort noise" (profile: supress-cng=true), the provider sends cn packets too. I tried with no luck variables such as: - send_silence_when_idle (with values "400" and "true") - bridge_generate_comfort_noise=true The rtp related params of my sofia profile are: rtp-timer-name=soft inbound-zrtp-passthru=false (don't use zrtp) rtp-timeout-sec=300 rtp-hold-timeout-sec=1800 inbound-late-negotiation=true NDLB-allow-bad-iananame=true supress-cng=true vad=none t38-passthru=true inbound-codec-prefs=PCMA,PCMU,G722,GSM outbound-codec-prefs=PCMA,PCMU,G722,GSM In the dialplan the only variables that I set related with rtp are: set inherit_codec=true export absolute_codec_string=${ep_codec_string} Thanks in advance -- Jose Fco. Irles Durá

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