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ClueCon 2011 Registration Open, Call For Speakers
by Michael Collins 22 Mar '11

22 Mar '11

Greetings fellow open source VoIP users and developers! We are happy to announce that ClueCon 2011 registration is now open. The event will be held August 9-11, 2011 at the Sofitel Hotel in downtown Chicago. Additionally, we are now accepting proposals for speaking topics. If you or your organization would like to speak at ClueCon this year please contact us with your ideas. Visit us at http://www.cluecon.com for additional details on registration, speaking, or sponsorship opportunities. Thank you to all the members of the open source telephony projects who have supported ClueCon over the years. We hope to see you all again this coming August! Michael S Collins ClueCon Team Member http://www.cluecon.com

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New FreeSWITCH Module: mod_ladspa
by Michael Collins 22 Mar '11

22 Mar '11

Hello all! The FreeSWITCH developers have added a cool new module: mod_ladspa. Check out the story: http://www.freeswitch.org/node/313 Have fun! -Michael

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No ringback tone, when calling
by Achim Stamm 18 Mar '11

18 Mar '11

No ringback tone, when calling Hello, i have the following problem: I use a phone on line port of LinkSys Spa 3102, which is registered with number 60 on freeswitch. I can hear a ringback tone, when I?m calling another registered phone for example 61 (see dialplan extension Call Voip). I cannot hear the ring tone on the phone connected to the Spa 3102, when I'm calling special number 70 leading to a self implemented freeswitch module (see dialplan Doing Something). This problem doesn't appear using X-Lite or an Siemens Voip telephone. My module does a preanswer as in the code shown below but something must be different - maybe I have to add somehting to the callhandling. Does anyone have an idea ? Greetings Achim Stamm Here is a snippet of my dialplan: ------------------------------------------------- <extension name="Doing Something"> <condition field="caller_id_number" expression="^6[0-9]$" /> <condition field="destination_number" expression="^7([0-9])$" > <action application="set" data="call_timeout=120"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="ignore_early_media=true"/> <action application="doingSomething" data="DoingSomething,5$1,${destination_number}"/> </condition> </extension> <extension name="Call Voip"> <condition field="destination_number" expression="^.*$" /> <condition field="caller_id_number" expression="6([0-9])"> <action application="bridge" data="sofia/internal/${destination_number}@${amtsleitung_5$1_ip_address}:5061" /> <action application="answer"/> </condition> </extension> ------------------------------------------------- Here is a snippet of my code: --------------------------------------------------- No ringback tone, when calling Hello, i have the following problem: I use a phone on line port of LinkSys Spa 3102, which is registered with number 60 on freeswitch. I can hear a ringback tone, when I?m calling another registered phone for example 61. I cannot hear the ring tone on the phone connected to the Spa 3102, when I'm calling special number 70 leading to a self implemented freeswitch module. This problem doesn't appear using X-Lite or an Siemens Voip telephone. My module does a preanswer as in the code shown below but something must be different - maybe I have to add somehting to the callhandling. Does anyone have an idea ? Greetings Achim Stamm Here is a snippet of my dialplan: ------------------------------------------------- <extension name="Doing Something"> <condition field="caller_id_number" expression="^6[0-9]$" /> <condition field="destination_number" expression="^7([0-9])$" > <action application="set" data="call_timeout=120"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="ignore_early_media=true"/> <action application="doingSomething" data="DoingSomething,5$1,${destination_number}"/> </condition> </extension> <extension name="Call Voip"> <condition field="destination_number" expression="^.*$" /> <condition field="caller_id_number" expression="6([0-9])"> <action application="bridge" data="sofia/internal/${destination_number}@${amtsleitung_5$1_ip_address}:5061" /> <action application="answer"/> </condition> </extension> ------------------------------------------------- Here is a snippet of my code: --------------------------------------------------- static switch_status_t doingSomething(switch_core_session_t *session, bool bWriteAudioLogfile, switch_input_args_t *args) { switch_codec_t codec = { 0 }; switch_status_t status; switch_frame_t *read_frame; switch_channel_t *channel = switch_core_session_get_channel(session); if (switch_channel_pre_answer(channel) != SWITCH_STATUS_SUCCESS) { return SWITCH_STATUS_FALSE; } while (switch_channel_ready(channel) ) { if (!process_read_write_frames(session,channel)) break; } switch_channel_hangup(channel,SWITCH_CAUSE_NORMAL_CLEARING); switch_log_printf(SWITCH_CHANNEL_LOG,SWITCH_LOG_INFO,"doingSomething\n"); return SWITCH_STATUS_SUCCESS; } --------------------------------------------------- --------------------------------------------------- -- Achim Stamm, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 04323897052 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis, Patrick Schmidt

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ready() function inside mod_spidermonkey returning False
by teldev 17 Mar '11

17 Mar '11

This issue appears to be an open JIRA, so we've added Comments to the JIRA but this post contains a few extra details as well. We have also analyzed the C source code and may have found the cause. http://jira.freeswitch.org/browse/FS-2966?focusedCommentId=22831#action_228… Here is our system environment: CentOS 5.5 x64 Using GIT Head from 3/1/2011 Linux ivr.anyco.corp 2.6.18-194.32.1.el5 #1 SMP Wed Jan 5 17:52:25 EST 2011 x86_64 x86_64 x86_64 GNU/Linux processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 15 model name : Intel(R) Xeon(R) CPU 5160 @ 3.00GHz stepping : 6 cpu MHz : 2999.999 cache size : 4096 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 2 apicid : 0 fpu : yes fpu_exception : yes cpuid level : 10 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc pni monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr lahf_lm bogomips : 5999.99 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 15 model name : Intel(R) Xeon(R) CPU 5160 @ 3.00GHz stepping : 6 cpu MHz : 2999.999 cache size : 4096 KB physical id : 0 siblings : 2 core id : 1 cpu cores : 2 apicid : 1 fpu : yes fpu_exception : yes cpuid level : 10 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc pni monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr lahf_lm bogomips : 5999.86 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: We are trying to originate a new call from an existing session from javascript using the session constuctor and then calling the ready() function. We have had intermittent call failures from calls generated by mod_spidermonkey. The following is a sequence of events that lead to failure of call originated from javascript interpreted by mod_spidermonkey. The following are from Freeswitch log, and my comments are on the end of the lines followed by // 2011-02-28 14:04:51.644009 [DEBUG] switch_ivr_originate.c:3408 Originate Resulted in Success: [sofia/external/5551212@10.10.10.28:5080] 2011-02-28 14:04:51.644009 [DEBUG] mod_spidermonkey.c:2885 (sofia/external/5551212@10.10.10.28:5080) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE //This occurs in the session constructor inside mod_spidermonkey 2011-02-28 14:04:51.644009 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/5551212@10.10.10.28:5080 [BREAK] 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[pupil_Init].[5].[5551212] // Our call to “ready()” returns before the running state changes 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[pupil_Init].[5].[] 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[pupil_Init].[5].[CS_SOFT_EXECUTE] // The state is already CS_SOFT_EXECUTE 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[pupil_Init].[5].[NONE] 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[pupil_Init].[6].[return false] 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[coach_NewSession].[pupil_Init].[failed] // But the return value of “ready()” is false 2011-02-28 14:04:51.644009 [DEBUG] fs_General.js:233 [debugmsg].[].[coach.js].[coach_NewSession].[7].[] 2011-02-28 14:04:51.644009 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/scripts/javascript/apps/public/audio/connection-failed.wav sample rate 16000 doesn't match requested rate 8000 2011-02-28 14:04:51.644009 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms 2011-02-28 14:04:51.645596 [DEBUG] switch_core_state_machine.c:320 (sofia/external/5551212@10.10.10.28:5080) Running State Change CS_SOFT_EXECUTE // The running state changed after “ready()” returns Looking inside the source code for “mod_spidemonkey”, it calls “switch_channel_wait_for_state_timeout” on line 2886, which is supposed to check for the “running state”, not the “state”. However, inside “switch_channel_wait_for_state_timeout” in switch_channel.c, it compares “running state” with “want_state” by a “>=”. Since CS_CONSUME_MEDIA > CS_SOFT_EXECUTE, this returns true before the running state actually changes. We cannot easily flip the “>=” check, either. There are enumeration values such as CS_NEW that are actually smaller than CS_SOFT_EXECUTE and should have precedence. Perhaps being in the state of “CS_CONSUME_MEDIA” is an unexpected condition to begin with? On line 15899 of “switch_channel.c”, it checks “!switch_channel_state_change_pending(channel)”. This check fails because the “running state” and the “state” are different. In other words, the state change is indeed pending. Because the transition time between CS_CONSUME_MEDIA going into CS_SOFT_EXECUTE can't be pre-determined, waiting for the completion of a call can make all calls take a long time to get answered. After running several tests the failure rate is somewhat unpredictable but fewer failures occur with a softphone like Kapanga (1 out of 4 fails) versus a hardware SIP phone like our Aastra 6739i SIP phone (3 out of 4 fails). Thanks in advance for your assistance. -- View this message in context: http://freeswitch-dev.995408.n3.nabble.com/ready-function-inside-mod-spider… Sent from the FreeSWITCH Dev mailing list archive at Nabble.com.

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How to read RTP package/data?
by Frankie Yiu 17 Mar '11

17 Mar '11

Hi there, I am new to FreeSwitch, and I have a question to ask. If I want to read the RTP package (data from a person picking up a phone and speak or an answer machine) in C++ or C#, what function should I call to get it? Currently I can make a call to a phone and play an audio. Thanks in advance. Frankie ____________________________________________________________________________________ Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/

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Local developer
by Stephen Picardo 16 Mar '11

16 Mar '11

Looking for local New England developer, to finish up Freeswitch project. Written in LUA and using MYSQL database with Sangoma 101D card. Original developer in Bankruptcy, and second individual can not seem to get things straighten out.

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FreeSWITCH Conference Call Today
by Michael Collins 16 Mar '11

16 Mar '11

Hello all! The agenda page for today is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_16 It's pretty light, so bring your questions and suggestions. -Michael

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Git Down
by Meftah Tayeb 12 Mar '11

12 Mar '11

FYI, i just try to pull git and is down this morning 10:00am gmt+1 thank you. -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

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FreeSWITCH Conference Call Followup - Recordings Online
by Michael Collins 09 Mar '11

09 Mar '11

FYI, We had a nice conference call with Jim Gettys and Dave Täht from the Bufferbloat project. I've made the recordings available in their usual spot: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls Use the links next to "FS_weekly_2011_03_09" to download the audio format of your choice. Thanks to Jim, Dave, and all those who participated today! -MC

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FreeSWITCH Conference Call Today: Special Presentation
by Michael Collins 09 Mar '11

09 Mar '11

Hello all! Today we are having a special presentation by Jim Gettys and Dave Taht of the Bufferbloat project. Please join us and learn more about this subject - you'll be glad you did. The official agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_09 Talk to you soon! -Michael

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