March 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] gsmopen and skypopen
by Giovanni Maruzzelli 28 Mar '11

28 Mar '11

Dear FreeSWITCHers, in the last weeks I've been too much busy, and I was not present in the community as I would have liked to be. I'll be much more present starting next week. I'm planning new features and more easy of installation for mod_skypopen. In the last month I saw some interest for gsmopen, that I was thinking to discontinue. This obviously is gratifying, and is pushing me to continue with gsmopen too, and make it more useful. Stay tuned starting next week ;). ciao, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

1 0

[Freeswitch-dev] Issue with in-band DTMF when using session:recordFile
by Daniel Casanova 24 Mar '11

24 Mar '11

I'm using Mod lua's session:recordFile to record audio to a file, but when working with in-band DTMF, the digit I set to stop the recording not only stops it but its tone is also recorded in the audio file (as a result, all my recorded files end having a beep sound at the end of the file). I tried the following code from http://wiki.freeswitch.org and it only happens when working with in-band DTMF. When working with RFC 2833 there are no beeping at all. function onInput(s, type, obj) if (type == "dtmf" and obj['digit'] == '#') then return "break"; end session:answer(); session:setInputCallback("onInput", ""); session:recordFile("/tmp/blah.wav", 30000, 10, 10); -- pressing # ends the recording session:streamFile("/tmp/blah.wav"); session:hangup(); Any help would be greatly appreciated. Thanks. Daniel

1 0

[Freeswitch-dev] How to route a single channel to multiple conference rooms...
by Jean-Pierre Poulin 23 Mar '11

23 Mar '11

Hello Freeswitch devs… First of all, congratulations on a solid telephony platform… I am particularly impressed with your modular architecture, the careful attention to thread safety, the clean API and the fact that the code can be built from both Linux & windows without a hitch. I need to be able to route a call to several conference rooms simultaneously. Can this be done in Freeswitch? The channel that needs to be multiplied to several conference rooms is input only (i.e. only the person’s voice is fed into the many conference rooms), and that person would not hear any output from any of the conference rooms. (For simplicity below, let’s call this source channel “Channel S”) Seeing that the public api implies that a given channel can only be present in one conference call at a time, I think this needs to be done by hacking the code… Studying the code for mod_conference, I am thinking of approaching this task in one of three ways: 1. Insert ‘Channel S’ into the list of conference participants in multiple conferences. 1. Create a normal conference #1 with channel S as a participant, and other conferences without channel S. 2. Extract the “conference_member” structure for channel S from conference #1 and copy this structure into the other conference rooms. 3. Trap deletion of “channel S” on conference #1 and duplicate the deletion to other conferences as needed. 4. QUESTION: Would issues between the timing between conference rooms or thread issues interfere? 2. Have ‘Channel S’ join a one person conference, record that conference, and instead of routing the outgoing sound packets to disk, patch “conference_play_file” in all conference rooms to insert the latest sound packet from channel S into each conference room. 1. QUESTION: This would only work if the sampling rate & timing is the same between record and “conference_play_file”. Is it? 3. Any other suggestion you might have… perhaps there is a non-public call I could invoke that could create this effect? Which of the above three do you think has the highest probability of working? Any ideas? Jean-Pierre

5 16

[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 23 Mar '11

23 Mar '11

Hey all, Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_23 We have a few items to discuss today, including the highlights from February's git commits. Also, we are going to have a discussion on some troubleshooting and data collection techniques. Talk to you soon! -MC

1 0

[Freeswitch-dev] How to dial out from an existing conference?
by Jean-Pierre Poulin 22 Mar '11

22 Mar '11

Hi all, I am attempting to initiate an outgoing call from a valid conference, and after over 10 hours of head scratching trying many things on two different builds I have not been able to do so. Has anyone been able to initiate calls from conference rooms to invite another extension to the conference? I have tried your suggested commands and they give me the same error messages as “conference dial” According to the docs, I should be able to initiate an outgoing call to a registered extension with a command like “conference 3000-192.168.0.11 dial sofia/internal/1001(a)192.168.0.xn--11-02t My setup is as follows: - Latest version of the latest Freeswtich installed on Windows 7 / 64 bit. (version reports "FreeSWITCH Version 1.0.head (git-cab1565 2010-05-18 10-42-16 -0400)") - All configuration settings are at their defaults, including the dialplan. - By registering an extension 1001 with X-lite on the main computer, and extension 1002 on another computer, both can call each other normally and join conferences together and everything works great. - Thinking this could be an X-lite issue, I have also registered extension 1004 on Bria on iPhone 4 and it can connect directly to the others, be connected by the other extensions and can connect fine to conference calls. For my application I need Freeswitch conferences to be able to dial out to other registed phones. After reading the docs, studying the dialplan, I have still been unable to do so after a half-day of trying various permutations and am currently grasping at straws… (I am currently compiling Freeswtich for Linux to see if it works there) I have also tried the following commands as you suggested: - originate sofia/internal/1004(a)192.168.0.11 &conference(3300-192.168.0.11) - originate sofia/internal/1004(a)192.168.0.11 &conference(3300-192.168.0.11@default) - originate sofia/internal/1004(a)192.168.0.11 conference:3300@default inline - originate sofia/internal/1004(a)192.168.0.11 conference:3300-192.168.0.11@default inline But in all cases to any registered phones on 3 machines get an error as “NO_USER_RESPONSE” and “Cannot Blind Transfer 1 Legged Calls” as the log entry below reveals. Should I try this whole thing in Freeswitch on Linux? Jean-Pierre ---------------------- Example log below: freeswitch@JPM> conference 3000-192.168.0.11 dial sofia/internal/1002(a)192.168.0.11 2011-03-18 17:43:14.407892 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1002(a)192.168.0.11 [0b7f1ab1-2251-4bfb-946e-e5fc639a09b9] 2011-03-18 17:43:14.410892 [NOTICE] switch_channel.c:675 New Channel sofia/internal/0000000000(a)192.168.0.11 [0a64217d-e1ff-4da8-a0b3-e84546f2e029] 2011-03-18 17:43:14.467895 [INFO] mod_dialplan_xml.c:331 Processing FreeSWITCH->1002 in context public 2011-03-18 17:43:14.471896 [ERR] sofia.c:5413 Cannot Blind Transfer 1 Legged calls Call Requested: result: [NO_USER_RESPONSE] 2011-03-18 17:43:14.472896 [NOTICE] sofia.c:4836 Hangup sofia/internal/1002(a)192.168.0.11 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2011-03-18 17:43:14.473896 [ERR] mod_conference.c:4597 Cannot create outgoing channel, cause: NO_USER_RESPONSE freeswitch@JPM> 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1188 Session 43 (sofia/internal/1002(a)192.168.0.11) Ended 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/1002(a)192.168.0.11 [CS_DESTROY] 2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:185 sofia/internal/0000000000(a)192.168.0.11 has executed the last dialplan instruction, hanging up. 2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/0000000000(a)192.168.0.11 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1188 Session 44 (sofia/internal/0000000000(a)192.168.0.11) Ended 2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/0000000000(a)192.168.0.11 [CS_DESTROY]

2 3

[Freeswitch-dev] sequence of channel events for bridge
by Ejay Greeves 22 Mar '11

22 Mar '11

I want to get the uuid of a bridge call. From which event can get the variable of the uuid? What is the sequence of channel events once a bridge command is executed, I captured a debug log and it did not contain CHANNEL BRIDGE even though I could hear that the call was connected to bleg

2 1

[Freeswitch-dev] 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

1 0

[Freeswitch-dev] 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

2 1

[Freeswitch-dev] 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

2 1

[Freeswitch-dev] 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.

2 1