From anthony.minessale at gmail.com Tue Mar 1 20:18:29 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Mar 2011 11:18:29 -0600 Subject: [Freeswitch-dev] [ANNOUNCEMENT] EVERYONE PLEASE READ Message-ID: Hi, I just want to give a reminder that we have a growing community and we want to put our resources to their best use. Here are some important things to consider. Many of you are already doing these things so if you already are, thank you, please help encourage others to follow your example. 1) We have a bug/issue tracker site using JIRA: http://jira.freeswitch.org This site is for reporting and discussing issues. Please use this tool to keep track of issues you have. Even if it's not a bug we can easily close it tagged "not a bug". Also please help out when you can by reviewing this site to see if you can provide answers to other people. 2) If you see someone complain about the WIKI please remind them that anyone can edit the WIKI and if it's wrong we can clean it up. It takes longer to complain about the wiki than it does to edit it to be correct or add the missing info. That is the idea behind wikis. 3) Please take the time to provide answers to any questions you may know when perusing the list. The more people who help the less work it is to keep everyone informed. 4) Take the time to test the latest development build. Its better to find bugs before we release than after. People tend to anchor themselves to the "official" releases and never try the BETA versions. The sad truth is the tagged release will begin to build a large list of known bugs starting 30 seconds after its tagged. You don't have to run HEAD in production every morning but you should stay up to speed with what's going on with the software and help to produce a stable release that truly is stable. 5) We will be branching 1.0 so it will slow down and taper off from upstream patches from HEAD. Anyone who wants to help maintain that via JIRA and GIT should let us know. Once we branch, most of our attention will still be focused on HEAD and only critical patches or bug fixes will be shared between 1.0 and 1.2 and we will rely on those interested in slower paced development to help keep that code base in order. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed Mar 2 19:40:13 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Mar 2011 08:40:13 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_02 We are going to have an update from Mark Crane on the FusionPBX project. Bring your questions! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110302/782e2994/attachment.html From msc at freeswitch.org Wed Mar 9 19:12:11 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Mar 2011 08:12:11 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today: Special Presentation Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110309/e6eaaf95/attachment.html From msc at freeswitch.org Thu Mar 10 01:24:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Mar 2011 14:24:50 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Followup - Recordings Online Message-ID: FYI, We had a nice conference call with Jim Gettys and Dave T?ht 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110309/aa443df4/attachment.html From msc at freeswitch.org Wed Mar 16 18:44:33 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Mar 2011 08:44:33 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110316/6a8b0424/attachment.html From tayeb.meftah at gmail.com Sat Mar 12 12:40:31 2011 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 12 Mar 2011 10:40:31 +0100 Subject: [Freeswitch-dev] Git Down Message-ID: <4D7B3F8F.7030102@gmail.com> FYI, i just try to pull git and is down this morning 10:00am gmt+1 thank you. -- Meftah Tayeb inum: +883510001288000 phone: +13477595883 From ustcorporation at yahoo.com Thu Mar 3 05:48:13 2011 From: ustcorporation at yahoo.com (teldev) Date: Wed, 2 Mar 2011 18:48:13 -0800 (PST) Subject: [Freeswitch-dev] ready() function inside mod_spidermonkey returning False Message-ID: <1299120493575-2622494.post@n3.nabble.com> 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_22831 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 at 10.10.10.28:5080] 2011-02-28 14:04:51.644009 [DEBUG] mod_spidermonkey.c:2885 (sofia/external/5551212 at 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 at 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 at 8000hz 1 channels 20ms 2011-02-28 14:04:51.645596 [DEBUG] switch_core_state_machine.c:320 (sofia/external/5551212 at 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-spidermonkey-returning-False-tp2622494p2622494.html Sent from the FreeSWITCH Dev mailing list archive at Nabble.com. From michelhabib at gmail.com Mon Mar 7 12:55:32 2011 From: michelhabib at gmail.com (Michel Habib) Date: Mon, 7 Mar 2011 11:55:32 +0200 Subject: [Freeswitch-dev] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem Message-ID: Hello All, I have MS OCS Speech Server 2007 [working correctly, as i can make SIP calls and use its ASR and TTS Services successfully] I am also using MRCP Connector from AumTech - which allows me to use ASR and TTS Services through an MRCP Client . Now, i am using Freeswitch mod unimrcp to use ASR and TTS. for TTS, I can successfully make the call, the Audio RTP of the TTS voice is transferred succesfully from Speech Server [through MRCP Connector] back to the Freeswitch Server. However, Freeswitch is not sending back the Audio RTP to the SIP client. for ASR, I can successfully define the grammar and start recognition, but the audio RTP sent to speech server [through MRCP Connector] is silent [empty]. I am suspecting something is wrong with the RTP Configuration - can you help me? Let me now if you need any specific logs/scripts/configuration? Thank you, Michel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110307/6de98d60/attachment-0001.html From stephen at picardogroup.com Thu Mar 17 01:44:38 2011 From: stephen at picardogroup.com (Stephen Picardo) Date: Wed, 16 Mar 2011 17:44:38 -0500 (CDT) Subject: [Freeswitch-dev] Local developer Message-ID: <1300315478.v2.mailanyonewebmail-267205@fuseweb2c> 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110316/211452f5/attachment.html From jeanpierrepoulin at yahoo.com Thu Mar 17 05:31:30 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Wed, 16 Mar 2011 19:31:30 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... Message-ID: <735336.10285.qm@web36807.mail.mud.yahoo.com> 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 From paul at cupis.co.uk Thu Mar 17 12:49:38 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 17 Mar 2011 09:49:38 +0000 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <735336.10285.qm@web36807.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> Message-ID: <4D81D932.9070801@cupis.co.uk> On 17/03/11 02:31, Jean-Pierre Poulin wrote: > I need to be able to route a call to several conference rooms simultaneously. > Can this be done in Freeswitch? Have you considered this model: Caller S calls into conference #1 Conference #2,#3,#4 are connected to conference #1. So you'd have conf#1 participant S participant conf#2 participant conf#3 participant conf#4 conf#2 participant L1 participant L2 participant L3 participant conf#1 conf#3 participant L4 participant L5 participant L6 participant conf#1 etc You'd need to ensure that the participants in conf#1 are all muted except for S to ensure feedback does not happen. Regards, From stamm at lyth.de Thu Mar 17 20:29:02 2011 From: stamm at lyth.de (Achim Stamm) Date: Thu, 17 Mar 2011 18:29:02 +0100 Subject: [Freeswitch-dev] No ringback tone, when calling Message-ID: <4D8244DE.3070506@lyth.de> 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: ------------------------------------------------- ------------------------------------------------- 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: ------------------------------------------------- ------------------------------------------------- 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 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis, Patrick Schmidt From jeanpierrepoulin at yahoo.com Thu Mar 17 20:31:37 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Thu, 17 Mar 2011 10:31:37 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <4D81D932.9070801@cupis.co.uk> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> Message-ID: <187004.51023.qm@web36802.mail.mud.yahoo.com> Hi Paul, thanks for your feedback and idea.? I think it?could work. I did not know conference rooms could be linked together by a channel. However... trying to go about understanding how to bridge two conference rooms, I'm having difficulty in finding the cli commands that could do this... could you give me a hint in the right direction? I would understand how to do this if I could create a one-legged call, park it, create another one-legged call, park it and use uuid_bridge or uuid_transfer to set both ends between two conference rooms to bridge them together, but the part I'm struggling with is during the call creation: the only type of calls I know how to make are to registered endpoints like a SIP phone, but in this case these have no value as the final endpoint for both legs have to be conference rooms. Put differently... do you know how to go about connecting two conference rooms together?? A few of the many silly things I've tried: - conference 3000-192.168.0.11 dial conference:3001-192.168.0.11 at default inline - conference 3000-192.168.0.11 dial conference:3001-192.168.0.11 - conference 3000-192.168.0.11 dial sofia/internal/3001-192.168.0.11- conference 3000-192.168.0.11 dial 3001-192.168.0.11 at default inline - conference 3000-192.168.0.11 dial 3001-192.168.0.11 Which basically all give errors that the recipient is either not registered, or 'cannot blind transfer to?1 legged call' Any ideas anyone? ? Jean-Pierre >Have you considered this model: >Caller S calls into conference #1 >Conference #2,#3,#4 are connected to conference #1. From mbodbg at gmx.net Thu Mar 17 20:37:34 2011 From: mbodbg at gmx.net (mbo) Date: Thu, 17 Mar 2011 18:37:34 +0100 Subject: [Freeswitch-dev] Calls are hanging in State CS_REPORTING when processing call via mod_event_socket In-Reply-To: <68103.31487.qm@web30504.mail.mud.yahoo.com> References: <68103.31487.qm@web30504.mail.mud.yahoo.com> Message-ID: After an upgrade to the latest freeswitch version (FreeSWITCH Version 1.0.head (git-9c40e8e 2011-03-16 20-50-32 -0500)) we have a problem that sessions are not properly closed after a call. To reproduce the error we can use the following dialplan: To simulate the server we can use nc. We start it with nc ?l ?p 8025 and then initiate a test call. On the nc command line we then type the following commands: connect myevents sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: hangup exit When we execute after the call ?show channels? on the freeswitch CLI, we can see that the channels are hanging in ?CS_REPORTING STATE?: 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 66e3c608-50b9-11e0-adad-db99c44ff3d1,inbound,2011-03-17 18:10:04,1300381804,FreeTDM/1:1/02214006783109,CS_REPORTING,,22155400323,,02214006783109,socket,127.0.0.1:8025 async full,XML,default,PCMA,8000,64000,PCMA,8000,64000,,VOXBACKCGN001,,,HANGUP,,,, a6cc3aac-50b9-11e0-adaf-db99c44ff3d1,inbound,2011-03-17 18:11:51,1300381911,FreeTDM/1:3/02214006783109,CS_REPORTING,,22155400323,,02214006783109,hangup,,XML,default,PCMA,8000,64000,PCMA,8000,64000,,VOXBACKCGN001,,,HANGUP,,,, If we do not send ?myevents? the channel is closed correctly after the hangup. We upgraded from FreeSWITCH Version 1.0.head (git-10d696e 2011-02-02 00-01-38 -0500), with this version we did not have that problem, the channel was closed correctly after the call. Any help would be appreciated Markus From anthony.minessale at gmail.com Thu Mar 17 20:44:49 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Mar 2011 12:44:49 -0500 Subject: [Freeswitch-dev] Calls are hanging in State CS_REPORTING when processing call via mod_event_socket In-Reply-To: References: <68103.31487.qm@web30504.mail.mud.yahoo.com> Message-ID: update to latest GIT and try not to report bugs to the mailing list. On Thu, Mar 17, 2011 at 12:37 PM, mbo wrote: > After an upgrade to the latest freeswitch version (FreeSWITCH Version 1.0.head (git-9c40e8e 2011-03-16 20-50-32 -0500)) we have a problem that sessions are not properly ?closed after a call. To reproduce the error we can use the following dialplan: > > ? > ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? > ? ? > > ?To simulate the server we can use nc. We start it with nc ?l ?p 8025 and then initiate a test call. On the nc command line we then type the following commands: > > > > connect > > > myevents > > sendmsg > call-command: execute > execute-app-name: answer > > sendmsg > call-command: execute > execute-app-name: hangup > > exit > > When we execute after the call ?show channels? on the freeswitch CLI, we can see that the channels are hanging in ?CS_REPORTING STATE?: > > 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 > 66e3c608-50b9-11e0-adad-db99c44ff3d1,inbound,2011-03-17 18:10:04,1300381804,FreeTDM/1:1/02214006783109,CS_REPORTING,,22155400323,,02214006783109,socket,127.0.0.1:8025 async full,XML,default,PCMA,8000,64000,PCMA,8000,64000,,VOXBACKCGN001,,,HANGUP,,,, > a6cc3aac-50b9-11e0-adaf-db99c44ff3d1,inbound,2011-03-17 18:11:51,1300381911,FreeTDM/1:3/02214006783109,CS_REPORTING,,22155400323,,02214006783109,hangup,,XML,default,PCMA,8000,64000,PCMA,8000,64000,,VOXBACKCGN001,,,HANGUP,,,, > > If we do not send ?myevents? the channel is closed correctly after the hangup. We upgraded from FreeSWITCH Version 1.0.head (git-10d696e 2011-02-02 00-01-38 -0500), with this version we did not have that problem, the channel was closed correctly after the call. > > > Any help would be appreciated > > > Markus > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From peter.olsson at visionutveckling.se Thu Mar 17 20:46:16 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 17 Mar 2011 18:46:16 +0100 Subject: [Freeswitch-dev] Calls are hanging in State CS_REPORTING when processing call via mod_event_socket In-Reply-To: References: <68103.31487.qm@web30504.mail.mud.yahoo.com>, Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58B2C49563@cooper> I think bug was just resolved in latest git (minutes ago). Update and try again. /Peter ________________________________________ Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] för mbo [mbodbg at gmx.net] Skickat: den 17 mars 2011 18:37 Till: freeswitch-dev at lists.freeswitch.org ?mne: [Freeswitch-dev] Calls are hanging in State CS_REPORTING when processing call via mod_event_socket After an upgrade to the latest freeswitch version (FreeSWITCH Version 1.0.head (git-9c40e8e 2011-03-16 20-50-32 -0500)) we have a problem that sessions are not properly closed after a call. To reproduce the error we can use the following dialplan: To simulate the server we can use nc. We start it with nc ?l ?p 8025 and then initiate a test call. On the nc command line we then type the following commands: connect myevents sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: hangup exit When we execute after the call ?show channels? on the freeswitch CLI, we can see that the channels are hanging in ?CS_REPORTING STATE?: 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 66e3c608-50b9-11e0-adad-db99c44ff3d1,inbound,2011-03-17 18:10:04,1300381804,FreeTDM/1:1/02214006783109,CS_REPORTING,,22155400323,,02214006783109,socket,127.0.0.1:8025 async full,XML,default,PCMA,8000,64000,PCMA,8000,64000,,VOXBACKCGN001,,,HANGUP,,,, a6cc3aac-50b9-11e0-adaf-db99c44ff3d1,inbound,2011-03-17 18:11:51,1300381911,FreeTDM/1:3/02214006783109,CS_REPORTING,,22155400323,,02214006783109,hangup,,XML,default,PCMA,8000,64000,PCMA,8000,64000,,VOXBACKCGN001,,,HANGUP,,,, If we do not send ?myevents? the channel is closed correctly after the hangup. We upgraded from FreeSWITCH Version 1.0.head (git-10d696e 2011-02-02 00-01-38 -0500), with this version we did not have that problem, the channel was closed correctly after the call. Any help would be appreciated Markus _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org !DSPAM:4d82471832765713110334! From paul at cupis.co.uk Thu Mar 17 21:04:08 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 17 Mar 2011 18:04:08 +0000 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <187004.51023.qm@web36802.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> Message-ID: <4D824D18.8010806@cupis.co.uk> On 17/03/11 17:31, Jean-Pierre Poulin wrote: > However... trying to go about understanding how to bridge two conference rooms, > I'm having difficulty in finding the cli commands that could do this... could > you give me a hint in the right direction? You want to do something like: conference conf#2 dial conf#1 So your dialplan allows people into conf#1 via, say: sofia/default/conference1 and conf#2 is number 1002 conference 1002 dial sofia/default/conference1 There are more details in the wiki: http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference These commands are not final, it does depend on your local configuration, also I've not taken into account the requirements to mute one leg of the joining calls. Regards, From anthony.minessale at gmail.com Fri Mar 18 01:33:39 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Mar 2011 17:33:39 -0500 Subject: [Freeswitch-dev] ready() function inside mod_spidermonkey returning False In-Reply-To: <1299120493575-2622494.post@n3.nabble.com> References: <1299120493575-2622494.post@n3.nabble.com> Message-ID: Can you retest with latest git 233d3164be4412aaaf8f9f42d8042e48279a018a or preferably the one from today? There is a line in originate now to wait for the state to settle before returning. This keeps your script from having access to the session before the state change is complete. What was happening is you were getting the session faster than the initial state change was complete during originate at which time it would go to sleep and do nothing unless someone transferred it etc but there is a delay from when you change a state and when that state actually happens. This should me moot in lastest GIT. On Wed, Mar 2, 2011 at 8:48 PM, teldev wrote: > 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_22831 > > 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 at 10.10.10.28:5080] > > 2011-02-28 14:04:51.644009 [DEBUG] mod_spidermonkey.c:2885 > (sofia/external/5551212 at 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 at 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 at 8000hz 1 channels 20ms > > 2011-02-28 14:04:51.645596 [DEBUG] switch_core_state_machine.c:320 > (sofia/external/5551212 at 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-spidermonkey-returning-False-tp2622494p2622494.html > Sent from the FreeSWITCH Dev mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jeanpierrepoulin at yahoo.com Fri Mar 18 02:21:57 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Thu, 17 Mar 2011 16:21:57 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <4D824D18.8010806@cupis.co.uk> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> Message-ID: <450146.58582.qm@web36807.mail.mud.yahoo.com> Hi Paul, thanks again for the help. Unfortunately,?these attempts to?connect a conference room?to a registered SIP phone all fail with the error "Cannot Blind Transfer 1 Legged calls" with a command like "conference 3000-192.168.0.11 dial sofia/internal/1001 at 192.168.0.11" even though conference room 3000 and extension 1001 are otherwise functional. (I'm using an untouched version of Freeswtich for Windows with the default configuration / dialplan) However, perhaps setting an extension?in the dialplan to route to a conference room could get me further... I will attempt to research?how that can be done. Thanks again! ?? Jean-Pierre ----- Original Message ---- >You want to do something like: >? conference conf#2 dial conf#1 >So your dialplan allows people into conf#1 via, say: >? sofia/default/conference1 >and conf#2 is number 1002 >? conference 1002 dial sofia/default/conference1 From paul at cupis.co.uk Fri Mar 18 11:12:05 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 18 Mar 2011 08:12:05 +0000 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <450146.58582.qm@web36807.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <450146.58582.qm@web36807.mail.mud.yahoo.com> Message-ID: <4D8313D5.7020206@cupis.co.uk> On 17/03/11 23:21, Jean-Pierre Poulin wrote: > However, perhaps setting an extension in the dialplan to route to a conference > room could get me further... I will attempt to research how that can be done. Okay, if you still can't get this working after trying the above with your dialplan, I suggest you post the query to the -users list where there will be more people who can help you (and how may have specific/exact examples for you). Regards, From msc at freeswitch.org Fri Mar 18 19:42:14 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Mar 2011 09:42:14 -0700 Subject: [Freeswitch-dev] No ringback tone, when calling In-Reply-To: <4D8244DE.3070506@lyth.de> References: <4D8244DE.3070506@lyth.de> Message-ID: Did you compare the siptraces on working vs. non-working calls? Any differences, like 180 vs. 183, etc.? -MC On Thu, Mar 17, 2011 at 10:29 AM, Achim Stamm wrote: > 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: > ------------------------------------------------- > > > > > > > data="DoingSomething,5$1,${destination_number}"/> > > > > > > > data="sofia/internal/${destination_number}@ > ${amtsleitung_5$1_ip_address}:5061" > /> > > > > ------------------------------------------------- > > 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: > ------------------------------------------------- > > > > > > > data="DoingSomething,5$1,${destination_number}"/> > > > > > > > data="sofia/internal/${destination_number}@ > ${amtsleitung_5$1_ip_address}:5061" > /> > > > > ------------------------------------------------- > > 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 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis, > Patrick Schmidt > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110318/86f81009/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 18 20:28:14 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Mar 2011 12:28:14 -0500 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <450146.58582.qm@web36807.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <450146.58582.qm@web36807.mail.mud.yahoo.com> Message-ID: Do you have a log of that because that makes no sense unless your phone is doing a refer. you can also do originate sofia/internal/foo at bar.com conference:xyz at default inline originate sofia/internal/foo at bar.com &conference(xyz at default) On Thu, Mar 17, 2011 at 6:21 PM, Jean-Pierre Poulin wrote: > Hi Paul, thanks again for the help. > > Unfortunately,?these attempts to?connect a conference room?to a registered SIP > phone all fail with the error "Cannot Blind Transfer 1 Legged calls" with a > command like "conference 3000-192.168.0.11 dial > sofia/internal/1001 at 192.168.0.11" even though conference room 3000 and extension > 1001 are otherwise functional. > > (I'm using an untouched version of Freeswtich for Windows with the default > configuration / dialplan) > > However, perhaps setting an extension?in the dialplan to route to a conference > room could get me further... I will attempt to research?how that can be done. > > Thanks again! > ?? Jean-Pierre > > > > ----- Original Message ---- >>You want to do something like: >>? conference conf#2 dial conf#1 >>So your dialplan allows people into conf#1 via, say: >>? sofia/default/conference1 >>and conf#2 is number 1002 >>? conference 1002 dial sofia/default/conference1 > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jeanpierrepoulin at yahoo.com Sat Mar 19 01:14:34 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Fri, 18 Mar 2011 15:14:34 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <450146.58582.qm@web36807.mail.mud.yahoo.com> Message-ID: <728055.17224.qm@web36806.mail.mud.yahoo.com> > Do you have a log of that because that makes no sense unless your phone is >doing a refer. > you can also do >originate sofia/internal/foo at bar.com conference:xyz at default inline >originate sofia/internal/foo at bar.com &conference(xyz at default) ---------------------------------- ? Hi Anthony, thanks for taking the time & congrats on fabulous Freeswitch!? J I have tried your suggested commands and they give me the same error messages as ?conference dial? 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? (Next attempt is to try the whole thing on Linux) 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 at 192.168.0.11? I have also tried the following commands as you suggested: - originate sofia/internal/1004 at 192.168.0.11 &conference(3300-192.168.0.11) - originate sofia/internal/1004 at 192.168.0.11 &conference(3300-192.168.0.11 at default) - originate sofia/internal/1004 at 192.168.0.11 conference:3300 at default inline - originate sofia/internal/1004 at 192.168.0.11 conference:3300-192.168.0.11 at 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 freeswitch at JPM> conference 3000-192.168.0.11 dial sofia/internal/1002 at 192.168.0.11 2011-03-18 17:43:14.407892 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1002 at 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 at 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 at 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 at JPM> 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1188 Session 43 (sofia/internal/1002 at 192.168.0.11) Ended 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/1002 at 192.168.0.11 [CS_DESTROY] 2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:185 sofia/internal/0000000000 at 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 at 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 at 192.168.0.11) Ended 2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] From msc at freeswitch.org Sat Mar 19 01:56:13 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Mar 2011 15:56:13 -0700 Subject: [Freeswitch-dev] New FreeSWITCH Module: mod_ladspa Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110318/a1b5f2a5/attachment.html From mike at jerris.com Sat Mar 19 18:53:05 2011 From: mike at jerris.com (Michael Jerris) Date: Sat, 19 Mar 2011 11:53:05 -0400 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <728055.17224.qm@web36806.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <450146.58582.qm@web36807.mail.mud.yahoo.com> <728055.17224.qm@web36806.mail.mud.yahoo.com> Message-ID: <12756E08-144F-46AB-B988-19C5515DCC84@jerris.com> linux vs windows won't make a difference here. Do you have forwarding turned on for the endpoint your having issues with? Mike On Mar 18, 2011, at 6:14 PM, Jean-Pierre Poulin wrote: >> Do you have a log of that because that makes no sense unless your phone is >> doing a refer. >> you can also do >> originate sofia/internal/foo at bar.com conference:xyz at default inline >> originate sofia/internal/foo at bar.com &conference(xyz at default) > > ---------------------------------- > > Hi Anthony, thanks for taking the time & congrats on fabulous Freeswitch! J > I have tried your suggested commands and they give me the same error messages as > ?conference dial? > 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? (Next attempt is to try the whole thing on Linux) > 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 at 192.168.0.11? > I have also tried the following commands as you suggested: > - originate sofia/internal/1004 at 192.168.0.11 &conference(3300-192.168.0.11) > - originate sofia/internal/1004 at 192.168.0.11 > &conference(3300-192.168.0.11 at default) > - originate sofia/internal/1004 at 192.168.0.11 conference:3300 at default inline > - originate sofia/internal/1004 at 192.168.0.11 > conference:3300-192.168.0.11 at 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 > freeswitch at JPM> conference 3000-192.168.0.11 dial > sofia/internal/1002 at 192.168.0.11 > 2011-03-18 17:43:14.407892 [NOTICE] switch_channel.c:675 New Channel > sofia/internal/1002 at 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 at 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 at 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 at JPM> 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1188 > Session 43 (sofia/internal/1002 at 192.168.0.11) Ended > 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1190 Close Channel > sofia/internal/1002 at 192.168.0.11 [CS_DESTROY] > 2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:185 > sofia/internal/0000000000 at 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 at 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 at 192.168.0.11) Ended > 2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1190 Close Channel > sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From jeanpierrepoulin at yahoo.com Sat Mar 19 19:40:38 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Sat, 19 Mar 2011 09:40:38 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <12756E08-144F-46AB-B988-19C5515DCC84@jerris.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <450146.58582.qm@web36807.mail.mud.yahoo.com> <728055.17224.qm@web36806.mail.mud.yahoo.com> <12756E08-144F-46AB-B988-19C5515DCC84@jerris.com> Message-ID: <743647.39041.qm@web36803.mail.mud.yahoo.com> > linux vs windows won't make a difference here.? Do you have forwarding turned >on for the endpoint your having issues with? Hi Mike, thanks for your input. As far as I know I don't have forwarding enabled on either my 2 X-lite extensions or my iPhone4's with Bria. I also tried compiling Freeswtich for Windows from source and it does the same thing (both have default configuration & dialplans) The only thing I can think of at this point is maybe it's a Windows 7 / 64 bit issues... grasping at straws!? Will try compiling Freeswitch on my Ubuntu VM to see if there is any difference. Thanks again for the help! ? Jean-Pierre From kris at kriskinc.com Tue Mar 22 17:34:15 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 22 Mar 2011 10:34:15 -0400 Subject: [Freeswitch-dev] New FreeSWITCH Module: mod_ladspa In-Reply-To: References: Message-ID: Cool. Would it be possible to do recqual-like stuff (in real time) with this module? On Fri, Mar 18, 2011 at 6:56 PM, Michael Collins wrote: > 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 > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Kristian Kielhofner From ejay.greeves at yahoo.com Fri Mar 18 14:19:46 2011 From: ejay.greeves at yahoo.com (Ejay Greeves) Date: Fri, 18 Mar 2011 11:19:46 +0000 (GMT) Subject: [Freeswitch-dev] sequence of channel events for bridge Message-ID: <997857.66328.qm@web132308.mail.ird.yahoo.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110318/2235690f/attachment-0001.html From ritzalam at gmail.com Mon Mar 21 21:58:22 2011 From: ritzalam at gmail.com (Richard Alam) Date: Mon, 21 Mar 2011 14:58:22 -0400 Subject: [Freeswitch-dev] Questions on implementing mod_nelly codec Message-ID: Hi, We've been trying to implement a mod_nelly codec but so far has been unsuccessful. We use a Nellymoser compatible codec submitted to ffmpeg (see June 16, 2008 entry of http://ffmpeg.org/). Our plan is to use Flash using Nellymoser to connect to a conference in FreeSWITCH. The hurdle seems to be that Nellymoser decodes 64 byte nelly audio to 512 byte L16. Nellymoser is 11-Khz 8-bit per sample with 32 ptime. However, L16 implementations are all 20 ptime which have uncompressed bytes in multiples of 320. This doesn't align with what mod_nelly expects when encoding (convert 512 bytes to 64 bytes) and decoding (64 bytes to 512 bytes). Here is the SWITCH_MODULE_LOAD_FUNCTION(mod_nelly_load) [http://pastebin.freeswitch.org/15764] now. We've played around with the samples per second (SPS), actual samples per second (ASPS), bits per second (BPS), and the ptime (PTIME) but with no luck. It's either we get bad very choppy audio, incompatible destination (when codecs can't match), or no audio at all. We've managed to get the "You will now be placed into the conference" audio played correctly but the audio from the caller is bad. Just the paying of wav file from FS is correct. My question is how does transcoding works in FS? Am I correct to assume decoding as Nelly->L16 and encoding as L16->Nelly? So basically the media flow for our use case is Nelly -> L16 --> mod_conference -> L16 -> Nelly. Do I need to make the frames align? In this case, Nelly expect 256 samples per frame while L16 has 160, 320, ... samples per frame. Hope I've articulated clearly what we are trying to accomplish. Suggestions and ideas on what to try out next is much appreciated. Thanks in advance. Richard -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton From marketing at cluecon.com Tue Mar 22 21:43:54 2011 From: marketing at cluecon.com (Michael Collins) Date: Tue, 22 Mar 2011 11:43:54 -0700 Subject: [Freeswitch-dev] ClueCon 2011 Registration Open, Call For Speakers Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110322/bfb6dd6b/attachment.html From jeanpierrepoulin at yahoo.com Tue Mar 22 22:14:47 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Tue, 22 Mar 2011 12:14:47 -0700 (PDT) Subject: [Freeswitch-dev] How to dial out from an existing conference? Message-ID: <818645.25999.qm@web36806.mail.mud.yahoo.com> 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 at 192.168.0.11? ? 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 at 192.168.0.11 &conference(3300-192.168.0.11) - originate sofia/internal/1004 at 192.168.0.11 &conference(3300-192.168.0.11 at default) - originate sofia/internal/1004 at 192.168.0.11 conference:3300 at default inline - originate sofia/internal/1004 at 192.168.0.11 conference:3300-192.168.0.11 at 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 at JPM> conference 3000-192.168.0.11 dial sofia/internal/1002 at 192.168.0.11 2011-03-18 17:43:14.407892 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1002 at 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 at 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 at 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 at JPM> 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1188 Session 43 (sofia/internal/1002 at 192.168.0.11) Ended 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/1002 at 192.168.0.11 [CS_DESTROY] 2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:185 sofia/internal/0000000000 at 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 at 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 at 192.168.0.11) Ended 2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] From anthony.minessale at gmail.com Wed Mar 23 00:19:17 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Mar 2011 16:19:17 -0500 Subject: [Freeswitch-dev] Questions on implementing mod_nelly codec In-Reply-To: References: Message-ID: Are you planning to contribute this module? On Mon, Mar 21, 2011 at 1:58 PM, Richard Alam wrote: > Hi, > > We've been trying to implement a mod_nelly codec but so far has been > unsuccessful. We use a Nellymoser compatible codec submitted to ffmpeg > (see June 16, 2008 entry of http://ffmpeg.org/). > > Our plan is to use Flash using Nellymoser to connect to a conference > in FreeSWITCH. > > The hurdle seems to be that Nellymoser decodes 64 byte nelly audio to > 512 byte L16. Nellymoser is 11-Khz 8-bit per sample with 32 ptime. > However, L16 implementations are all 20 ptime > which have uncompressed bytes in multiples of 320. This doesn't align > with what mod_nelly expects when encoding (convert 512 bytes to 64 > bytes) and decoding (64 bytes to 512 bytes). > > Here is the SWITCH_MODULE_LOAD_FUNCTION(mod_nelly_load) > [http://pastebin.freeswitch.org/15764] now. We've played around with > the samples per second (SPS), actual samples per second (ASPS), > bits per second (BPS), and the ptime (PTIME) but with no luck. It's > either we get bad very choppy audio, incompatible destination (when > codecs can't match), or no audio at all. We've managed to get the > "You will now be placed into the conference" audio played correctly > but the audio from the caller is bad. Just the paying of wav file from > FS is correct. > > My question is how does transcoding works in FS? Am I correct to > assume decoding as Nelly->L16 and encoding as L16->Nelly? > So basically the media flow for our use case is Nelly -> L16 --> > mod_conference -> L16 -> Nelly. > > Do I need to make the frames align? In this case, Nelly expect 256 > samples per frame while L16 has 160, 320, ... samples per frame. > > Hope I've articulated clearly what we are trying to accomplish. > > Suggestions and ideas on what to try out next is much appreciated. > > Thanks in advance. > > Richard > > > -- > --- > BigBlueButton > http://www.bigbluebutton.org > http://code.google.com/p/bigbluebutton > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed Mar 23 01:46:38 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Mar 2011 15:46:38 -0700 Subject: [Freeswitch-dev] How to dial out from an existing conference? In-Reply-To: <818645.25999.qm@web36806.mail.mud.yahoo.com> References: <818645.25999.qm@web36806.mail.mud.yahoo.com> Message-ID: You are gonna kick yourself when you hear the answer... You don't dial this: sofia/internal/1001 at 192.168.0.11 <1001 at 192.168.0.11>You dial one of these: sofia/internal/1001%192.168.0.11 <1001 at 192.168.0.11> <1001 at 192.168.0.11>-or- user/1001 See this wiki page about the anatomy of a sofia dialstring: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings I didn't catch this at first because I just naturally did the "user/xxxx" syntax. (I always use the "user/xxxx" syntax when calling a registered user on my own box, which it appears you are also doing with user 1001.) Keep in mind that of 1001 is not a registered user then you can't use "user/1001"... -MC On Tue, Mar 22, 2011 at 12:14 PM, Jean-Pierre Poulin < jeanpierrepoulin at yahoo.com> wrote: > 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 at 192.168.0.11? > > > 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 at 192.168.0.11&conference(3300-192.168.0.11) > - originate sofia/internal/1004 at 192.168.0.11 > &conference(3300-192.168.0.11 at default) > - originate sofia/internal/1004 at 192.168.0.11 conference:3300 at defaultinline > - originate sofia/internal/1004 at 192.168.0.11 > conference:3300-192.168.0.11 at 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 at JPM> conference 3000-192.168.0.11 dial > sofia/internal/1002 at 192.168.0.11 > 2011-03-18 17:43:14.407892 [NOTICE] switch_channel.c:675 New Channel > sofia/internal/1002 at 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 at 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 at 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 at JPM> 2011-03-18 17:43:14.544900 [NOTICE] > switch_core_session.c:1188 > Session 43 (sofia/internal/1002 at 192.168.0.11) Ended > 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1190 Close > Channel > sofia/internal/1002 at 192.168.0.11 [CS_DESTROY] > 2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:185 > sofia/internal/0000000000 at 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 at 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 at 192.168.0.11) Ended > 2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1190 Close > Channel > sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110322/dbe0e176/attachment.html From msc at freeswitch.org Wed Mar 23 01:52:49 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Mar 2011 15:52:49 -0700 Subject: [Freeswitch-dev] sequence of channel events for bridge In-Reply-To: <997857.66328.qm@web132308.mail.ird.yahoo.com> References: <997857.66328.qm@web132308.mail.ird.yahoo.com> Message-ID: Make sure that you are actually receiving events. You can go to fs_cli and do this: /log 0 /event plain all /filter Event-Name CHANNEL_BRIDGE Then, make a call that will do a bridge. Near the end of the event you should have "variable_signal_bond" which has the b-leg's uuid. -MC On Fri, Mar 18, 2011 at 4:19 AM, Ejay Greeves wrote: > 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 > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110322/849f7928/attachment-0001.html From jeanpierrepoulin at yahoo.com Wed Mar 23 01:58:48 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Tue, 22 Mar 2011 15:58:48 -0700 (PDT) Subject: [Freeswitch-dev] How to dial out from an existing conference? In-Reply-To: References: <818645.25999.qm@web36806.mail.mud.yahoo.com> Message-ID: <121245.73479.qm@web36803.mail.mud.yahoo.com> Hi Michael... (I am indeed?kicking myself?at every word as I write this email) I works!? Many thanks for taking the time... this has stumped?me for days... the error message about 'one legged calls' really threw me off on?that one! Thanks again for taking the time and?have a great day! ?Jean-Pierre ________________________________ From: Michael Collins To: freeswitch-dev at lists.freeswitch.org Sent: Tue, March 22, 2011 6:46:38 PM Subject: Re: [Freeswitch-dev] How to dial out from an existing conference? You are gonna kick yourself when you hear the answer... You don't dial this:? sofia/internal/1001 at 192.168.0.11 You dial one of these: sofia/internal/1001%192.168.0.11 -or- user/1001 See this wiki page about the anatomy of a sofia dialstring: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings I didn't catch this at first because I just naturally did the "user/xxxx" syntax. (I always use the "user/xxxx" syntax when calling a registered user on my own box, which it appears you are also doing with user 1001.) Keep in mind that of 1001 is not a registered user then you can't use "user/1001"... -MC On Tue, Mar 22, 2011 at 12:14 PM, Jean-Pierre Poulin wrote: 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 at 192.168.0.11? > >? >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 at 192.168.0.11 &conference(3300-192.168.0.11) >- originate sofia/internal/1004 at 192.168.0.11 >&conference(3300-192.168.0.11 at default) >- originate sofia/internal/1004 at 192.168.0.11 conference:3300 at default inline >- originate sofia/internal/1004 at 192.168.0.11 >conference:3300-192.168.0.11 at 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 at JPM> conference 3000-192.168.0.11 dial >sofia/internal/1002 at 192.168.0.11 >2011-03-18 17:43:14.407892 [NOTICE] switch_channel.c:675 New Channel >sofia/internal/1002 at 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 at 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 at 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 at JPM> 2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1188 >Session 43 (sofia/internal/1002 at 192.168.0.11) Ended >2011-03-18 17:43:14.544900 [NOTICE] switch_core_session.c:1190 Close Channel >sofia/internal/1002 at 192.168.0.11 [CS_DESTROY] >2011-03-18 17:43:14.570901 [NOTICE] switch_core_state_machine.c:185 >sofia/internal/0000000000 at 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 at 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 at 192.168.0.11) Ended >2011-03-18 17:43:14.572901 [NOTICE] switch_core_session.c:1190 Close Channel >sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] > > > >_______________________________________________ >FreeSWITCH-dev mailing list >FreeSWITCH-dev at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110322/6b9f8be9/attachment.html From msc at freeswitch.org Wed Mar 23 02:15:41 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Mar 2011 16:15:41 -0700 Subject: [Freeswitch-dev] How to dial out from an existing conference? In-Reply-To: <121245.73479.qm@web36803.mail.mud.yahoo.com> References: <818645.25999.qm@web36806.mail.mud.yahoo.com> <121245.73479.qm@web36803.mail.mud.yahoo.com> Message-ID: On Tue, Mar 22, 2011 at 3:58 PM, Jean-Pierre Poulin < jeanpierrepoulin at yahoo.com> wrote: > Hi Michael... > > (I am indeed kicking myself at every word as I write this email) > At least you won't forget this one any time soon! Also, you might be more sensitive to these issues when others have them and you'll know what to tell them. ;) -MC > > I works! Many thanks for taking the time... this has stumped me for > days... the error message about 'one legged calls' really threw me off > on that one! > > Thanks again for taking the time and have a great day! > Bien sur et de rien! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110322/596ebb32/attachment.html From msc at freeswitch.org Wed Mar 23 19:27:54 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Mar 2011 09:27:54 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110323/9ad90657/attachment.html From jeanpierrepoulin at yahoo.com Wed Mar 23 22:06:57 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Wed, 23 Mar 2011 12:06:57 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <4D824D18.8010806@cupis.co.uk> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> Message-ID: <940325.64326.qm@web36803.mail.mud.yahoo.com> > You want to do something like: >? conference conf#2 dial conf#1 >So your dialplan allows people into conf#1 via, say: >? sofia/default/conference1 >and conf#2 is number 1002 >? conference 1002 dial sofia/default/conference1 Hi Paul, thanks again for the hint in this direction, but after much research I just can't connect two conference rooms together. My best effort thus far involves registering extensions 3400..3499 in a similar way conferences 3000..3099 are done ??? ????? ??????? ??????? ????? ??? And to insert a new user in conf/directory/default for file 3400.xml so that user / conference 3400 can be seen as a registered user. ? ??? ????? ????? ??? ??? ????? ????? ????? ????? ????? ????? ????? ????? ??? ? Unforunately, when I call out with a cli command like "conference 3001-192.168.0.11 dial user/3400" I get 2011-03-23 14:56:46.770577 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] Call Requested: result: [USER_NOT_REGISTERED] 2011-03-23 14:56:46.770577 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2011-03-23 14:56:46.770577 [ERR] mod_conference.c:4597 Cannot create outgoing channel, cause: USER_NOT_REGISTERED (The above command works fine if I call a registered extension like 1001...? In other words, no endpoint appears to be present / answering for 3400 even though I have placed an "answer" application as done for the other conferences. (The same error appears even if I register two SIP phones to conference rooms 3000 and 3400) Has linking two conference rooms ever been done? Thanks for any hint you can offer... I have no idea what to try next and so far the best path forward looks like hacking the code. ?? Jean-Pierre From anthony.minessale at gmail.com Wed Mar 23 22:46:25 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Mar 2011 14:46:25 -0500 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <940325.64326.qm@web36803.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <940325.64326.qm@web36803.mail.mud.yahoo.com> Message-ID: yes I can do it effortlessly 100 times in a row from many boxes. You need to of course make sure the url you are dialing in the originate actually is reachable by FS. On Wed, Mar 23, 2011 at 2:06 PM, Jean-Pierre Poulin wrote: >> You want to do something like: >>? conference conf#2 dial conf#1 >>So your dialplan allows people into conf#1 via, say: >>? sofia/default/conference1 >>and conf#2 is number 1002 >>? conference 1002 dial sofia/default/conference1 > > Hi Paul, thanks again for the hint in this direction, but after much research I > just can't connect two conference rooms together. > > My best effort thus far involves registering extensions 3400..3499 in a similar > way conferences 3000..3099 are done > > ??? > ????? > ??????? > ??????? > ????? > ??? > > And to insert a new user in conf/directory/default for file 3400.xml so that > user / conference 3400 can be seen as a registered user. > > > ? > ??? > ????? > ????? > ??? > ??? > ????? > ????? > ????? > ????? > ????? > ????? value="$${outbound_caller_name}"/> > ????? value="$${outbound_caller_id}"/> > ????? > ??? > ? > > > Unforunately, when I call out with a cli command like "conference > 3001-192.168.0.11 dial user/3400" I get > > 2011-03-23 14:56:46.770577 [ERR] switch_ivr_originate.c:2493 Cannot create > outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > Call Requested: result: [USER_NOT_REGISTERED] > 2011-03-23 14:56:46.770577 [ERR] switch_ivr_originate.c:2493 Cannot create > outgoing channel of type [user] cause: [USER_NOT_REGISTERED] > 2011-03-23 14:56:46.770577 [ERR] mod_conference.c:4597 Cannot create outgoing > channel, cause: USER_NOT_REGISTERED > > (The above command works fine if I call a registered extension like 1001...? In > other words, no endpoint appears to be present / answering for 3400 even though > I have placed an "answer" application as done for the other conferences. > > (The same error appears even if I register two SIP phones to conference rooms > 3000 and 3400) > > Has linking two conference rooms ever been done? > > Thanks for any hint you can offer... I have no idea what to try next and so far > the best path forward looks like hacking the code. > > ?? Jean-Pierre > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jeanpierrepoulin at yahoo.com Wed Mar 23 23:32:07 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Wed, 23 Mar 2011 13:32:07 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <940325.64326.qm@web36803.mail.mud.yahoo.com> Message-ID: <769463.98519.qm@web36804.mail.mud.yahoo.com> >yes I can do it effortlessly 100 times in a row from many boxes. >You need to of course make sure the url you are dialing in the >originate actually is reachable by FS. Hi Anthony, thanks again for your time.? Knowing this is known to work is exactly what I needed at this time to keep going along this branch. In this particular dialplan (exact same as default dialplan + user 3400.xml added in user directory and dialplan/default.xml opening extension 3400 toward application "conference" as per previous message), I can call extension 3400 from any SIP phone so it is reachable by FS, but thus far the call "conference 3000-192.168.0.11 dial user/3400" simply fails with NOT_REGISTERED even if there is a participant in conference 3400 and conference 3400 is otherwise fully functional. Would you be so kind as telling me what CLI command you use when you want to bridge two conference rooms together? A thousand thanks for any hint you can offer!! ? Jean-Pierre From anthony.minessale at gmail.com Thu Mar 24 00:26:02 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Mar 2011 16:26:02 -0500 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <769463.98519.qm@web36804.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <940325.64326.qm@web36803.mail.mud.yahoo.com> <769463.98519.qm@web36804.mail.mud.yahoo.com> Message-ID: user/3400 would be a phone registered to ext 3400 not a magic way to call extension 3400 if you are trying to call another conference on the same box you need to loop the call over sip. try: conference 3000-192.168.0.11 dial sofia/internal/3400 at 1.2.3.4 replacing 1.2.3.4 with the ip or hostname of your box. On Wed, Mar 23, 2011 at 3:32 PM, Jean-Pierre Poulin wrote: >>yes I can do it effortlessly 100 times in a row from many boxes. >>You need to of course make sure the url you are dialing in the >>originate actually is reachable by FS. > > Hi Anthony, thanks again for your time.? Knowing this is known to work is > exactly what I needed at this time to keep going along this branch. > > In this particular dialplan (exact same as default dialplan + user 3400.xml > added in user directory and dialplan/default.xml opening extension 3400 toward > application "conference" as per previous message), I can call extension 3400 > from any SIP phone so it is reachable by FS, but thus far the call "conference > 3000-192.168.0.11 dial user/3400" simply fails with NOT_REGISTERED even if there > is a participant in conference 3400 and conference 3400 is otherwise fully > functional. > > Would you be so kind as telling me what CLI command you use when you want to > bridge two conference rooms together? > > A thousand thanks for any hint you can offer!! > > ? Jean-Pierre > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jeanpierrepoulin at yahoo.com Thu Mar 24 01:29:09 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Wed, 23 Mar 2011 15:29:09 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <940325.64326.qm@web36803.mail.mud.yahoo.com> <769463.98519.qm@web36804.mail.mud.yahoo.com> Message-ID: <515105.59158.qm@web36801.mail.mud.yahoo.com> > if you are trying to call another conference on the same box you need > to loop the call over sip. try: > conference 3000-192.168.0.11 dial sofia/internal/3400 at 1.2.3.4 Hi Anthony, sorry to bother you with this again, but I've tried your suggested command on three different builds (win binary, win from source, linux from source) all with?out-of-the-box dialplan & settings and they all return the 'cannot blind transfer 1 legged calls'. Log of my command attempting to connect conference 3000 and 3001 below:? (My IP address is 192.168.0.11) conference 3000-192.168.0.11 dial sofia/internal/3001 at 192.168.0.11 2011-03-23 18:23:42.298705 [NOTICE] switch_channel.c:812 New Channel sofia/internal/3001 at 192.168.0.11 [78b3b6d8-38ef-420d-ba88-72fcb3a1b61c] 2011-03-23 18:23:42.302705 [NOTICE] switch_channel.c:812 New Channel sofia/internal/0000000000 at 192.168.0.11 [079cf467-1db2-4f0b-b6db-06a0899600ea] 2011-03-23 18:23:42.307706 [INFO] mod_dialplan_xml.c:331 Processing FreeSWITCH <0000000000>->3001 in context public 2011-03-23 18:23:42.312706 [ERR] sofia.c:5906 Cannot Blind Transfer 1 Legged calls Call Requested: result: [NO_USER_RESPONSE] 2011-03-23 18:23:42.314706 [NOTICE] sofia.c:5323 Hangup sofia/internal/3001 at 192.168.0.11 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2011-03-23 18:23:42.314706 [ERR] mod_conference.c:5021 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2011-03-23 18:23:42.318706 [NOTICE] switch_core_session.c:1306 Session 9 (sofia/internal/3001 at 192.168.0.11) Ended 2011-03-23 18:23:42.318706 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/3001 at 192.168.0.11 [CS_DESTROY] freeswitch at JPM> 2011-03-23 18:23:42.412712 [NOTICE] switch_core_state_machine.c:189 sofia/internal/0000000000 at 192.168.0.11 has executed the last dialplan instruction, hanging up. 2011-03-23 18:23:42.412712 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/0000000000 at 192.168.0.11 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-23 18:23:42.422712 [NOTICE] switch_core_session.c:1306 Session 10 (sofia/internal/0000000000 at 192.168.0.11) Ended 2011-03-23 18:23:42.422712 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] From msc at freeswitch.org Thu Mar 24 02:19:56 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Mar 2011 16:19:56 -0700 Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: <515105.59158.qm@web36801.mail.mud.yahoo.com> References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <940325.64326.qm@web36803.mail.mud.yahoo.com> <769463.98519.qm@web36804.mail.mud.yahoo.com> <515105.59158.qm@web36801.mail.mud.yahoo.com> Message-ID: I hate to say it, but what happens when you try this? conference 3000-192.168.0.11 dial loopback/3001 I'm just curious. We aren't fond of the crazy things people try to do with loopback but I just want to know if this connects your two conferences. (It did on my system.) -MC On Wed, Mar 23, 2011 at 3:29 PM, Jean-Pierre Poulin < jeanpierrepoulin at yahoo.com> wrote: > > if you are trying to call another conference on the same box you need > > to loop the call over sip. try: > > conference 3000-192.168.0.11 dial sofia/internal/3400 at 1.2.3.4 > > Hi Anthony, sorry to bother you with this again, but I've tried your > suggested > command on three different builds (win binary, win from source, linux from > source) all with out-of-the-box dialplan & settings and they all return the > 'cannot blind transfer 1 legged calls'. > > Log of my command attempting to connect conference 3000 and 3001 below: > (My IP > address is 192.168.0.11) > > conference 3000-192.168.0.11 dial sofia/internal/3001 at 192.168.0.11 > > 2011-03-23 18:23:42.298705 [NOTICE] switch_channel.c:812 New Channel > sofia/internal/3001 at 192.168.0.11 [78b3b6d8-38ef-420d-ba88-72fcb3a1b61c] > 2011-03-23 18:23:42.302705 [NOTICE] switch_channel.c:812 New Channel > sofia/internal/0000000000 at 192.168.0.11[079cf467-1db2-4f0b-b6db-06a0899600ea] > 2011-03-23 18:23:42.307706 [INFO] mod_dialplan_xml.c:331 Processing > FreeSWITCH > <0000000000>->3001 in context public > 2011-03-23 18:23:42.312706 [ERR] sofia.c:5906 Cannot Blind Transfer 1 > Legged > calls > Call Requested: result: [NO_USER_RESPONSE] > 2011-03-23 18:23:42.314706 [NOTICE] sofia.c:5323 Hangup > sofia/internal/3001 at 192.168.0.11 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2011-03-23 18:23:42.314706 [ERR] mod_conference.c:5021 Cannot create > outgoing > channel, cause: NO_USER_RESPONSE > 2011-03-23 18:23:42.318706 [NOTICE] switch_core_session.c:1306 Session 9 > (sofia/internal/3001 at 192.168.0.11) Ended > 2011-03-23 18:23:42.318706 [NOTICE] switch_core_session.c:1308 Close > Channel > sofia/internal/3001 at 192.168.0.11 [CS_DESTROY] > freeswitch at JPM> 2011-03-23 18:23:42.412712 [NOTICE] > switch_core_state_machine.c:189 sofia/internal/0000000000 at 192.168.0.11 has > executed the last dialplan instruction, hanging up. > 2011-03-23 18:23:42.412712 [NOTICE] switch_core_state_machine.c:191 Hangup > sofia/internal/0000000000 at 192.168.0.11 [CS_EXECUTE] [NORMAL_CLEARING] > 2011-03-23 18:23:42.422712 [NOTICE] switch_core_session.c:1306 Session 10 > (sofia/internal/0000000000 at 192.168.0.11) Ended > 2011-03-23 18:23:42.422712 [NOTICE] switch_core_session.c:1308 Close > Channel > sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110323/9c643fa5/attachment.html From jeanpierrepoulin at yahoo.com Thu Mar 24 02:48:27 2011 From: jeanpierrepoulin at yahoo.com (Jean-Pierre Poulin) Date: Wed, 23 Mar 2011 16:48:27 -0700 (PDT) Subject: [Freeswitch-dev] How to route a single channel to multiple conference rooms... In-Reply-To: References: <735336.10285.qm@web36807.mail.mud.yahoo.com> <4D81D932.9070801@cupis.co.uk> <187004.51023.qm@web36802.mail.mud.yahoo.com> <4D824D18.8010806@cupis.co.uk> <940325.64326.qm@web36803.mail.mud.yahoo.com> <769463.98519.qm@web36804.mail.mud.yahoo.com> <515105.59158.qm@web36801.mail.mud.yahoo.com> Message-ID: <454481.77920.qm@web36808.mail.mud.yahoo.com> Ha!? That "conference 3000-192.168.0.11 dial loopback/3001" command did the trick! For the benefit of people later on looking on how to connect two conference rooms together, starting from out-of-the-box FreeSwitch you can connect any conference between 3000..3399 with a command like: conference 3000-192.168.0.11 dial loopback/3001 Once this is done, connect one SIP phone to conference 3000 and the other to 3001, and both SIP phones should magically be able to hear each other. You should be able to verify on both conferences that they are connected with a command like "conference 3000-192.168.0.11 list".? In the above-described situation this gives: 3;sofia/internal/1001 at 192.168.0.11;c6d34e33-5fcd-4a83-b049-f10934738a51;Jean-Pierre;1001;hear|speak;0;0;300 2;loopback/3001-a;1df73907-5eac-479e-bcda-610fb60e9364;;0000000000;hear|speak|talking|floor;0;0;300 Thanks a million to everyone who participated toward this answer... you?guys rock!? :) ??? Jean-Pierre P.S. Interestingly enough, I still hear music on hold in this situation... (not a big deal for me as I intended to switch it off anyways) ________________________________ From: Michael Collins To: freeswitch-dev at lists.freeswitch.org Sent: Wed, March 23, 2011 7:19:56 PM Subject: Re: [Freeswitch-dev] How to route a single channel to multiple conference rooms... I hate to say it, but what happens when you try this? conference 3000-192.168.0.11 dial loopback/3001 I'm just curious. We aren't fond of the crazy things people try to do with loopback but I just want to know if this connects your two conferences. (It did on my system.) -MC On Wed, Mar 23, 2011 at 3:29 PM, Jean-Pierre Poulin wrote: > if you are trying to call another conference on the same box you need >> to loop the call over sip. try: >> conference 3000-192.168.0.11 dial sofia/internal/3400 at 1.2.3.4 > >Hi Anthony, sorry to bother you with this again, but I've tried your suggested >command on three different builds (win binary, win from source, linux from >source) all with?out-of-the-box dialplan & settings and they all return the >'cannot blind transfer 1 legged calls'. > >Log of my command attempting to connect conference 3000 and 3001 below:? (My IP >address is 192.168.0.11) > >conference 3000-192.168.0.11 dial sofia/internal/3001 at 192.168.0.11 > >2011-03-23 18:23:42.298705 [NOTICE] switch_channel.c:812 New Channel >sofia/internal/3001 at 192.168.0.11 [78b3b6d8-38ef-420d-ba88-72fcb3a1b61c] >2011-03-23 18:23:42.302705 [NOTICE] switch_channel.c:812 New Channel >sofia/internal/0000000000 at 192.168.0.11 [079cf467-1db2-4f0b-b6db-06a0899600ea] >2011-03-23 18:23:42.307706 [INFO] mod_dialplan_xml.c:331 Processing FreeSWITCH ><0000000000>->3001 in context public >2011-03-23 18:23:42.312706 [ERR] sofia.c:5906 Cannot Blind Transfer 1 Legged >calls >Call Requested: result: [NO_USER_RESPONSE] >2011-03-23 18:23:42.314706 [NOTICE] sofia.c:5323 Hangup >sofia/internal/3001 at 192.168.0.11 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] >2011-03-23 18:23:42.314706 [ERR] mod_conference.c:5021 Cannot create outgoing >channel, cause: NO_USER_RESPONSE >2011-03-23 18:23:42.318706 [NOTICE] switch_core_session.c:1306 Session 9 >(sofia/internal/3001 at 192.168.0.11) Ended >2011-03-23 18:23:42.318706 [NOTICE] switch_core_session.c:1308 Close Channel >sofia/internal/3001 at 192.168.0.11 [CS_DESTROY] >freeswitch at JPM> 2011-03-23 18:23:42.412712 [NOTICE] >switch_core_state_machine.c:189 sofia/internal/0000000000 at 192.168.0.11 has >executed the last dialplan instruction, hanging up. >2011-03-23 18:23:42.412712 [NOTICE] switch_core_state_machine.c:191 Hangup >sofia/internal/0000000000 at 192.168.0.11 [CS_EXECUTE] [NORMAL_CLEARING] >2011-03-23 18:23:42.422712 [NOTICE] switch_core_session.c:1306 Session 10 >(sofia/internal/0000000000 at 192.168.0.11) Ended >2011-03-23 18:23:42.422712 [NOTICE] switch_core_session.c:1308 Close Channel >sofia/internal/0000000000 at 192.168.0.11 [CS_DESTROY] > > > >_______________________________________________ >FreeSWITCH-dev mailing list >FreeSWITCH-dev at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110323/b0355100/attachment-0001.html From gmaruzz at gmail.com Mon Mar 28 17:06:24 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 28 Mar 2011 15:06:24 +0200 Subject: [Freeswitch-dev] gsmopen and skypopen Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110328/216b6760/attachment.html From aroumie at yahoo.com Tue Mar 29 01:45:41 2011 From: aroumie at yahoo.com (Ali R.) Date: Mon, 28 Mar 2011 14:45:41 -0700 (PDT) Subject: [Freeswitch-dev] mod_socket Message-ID: <293813.9658.qm@web120604.mail.ne1.yahoo.com> Hi everyone I think my issue could be fixed with mod_xml_curl but I'm using FS purely through sockets (inbound mode) When I get the park event, I execute some biz logic.? Leg's A source IP plays a role in this logic so based on the value of the IP (a big pool of allowed IP addresses is generated dynamically and changes very fast so I have a logic to query this pool) all good so far. However, if the incoming channel?s source IP is not allowed I would still want to answer and continue on but I must challenge it with a user name and password that I have a logic to retrieve.? Any thoughts on how should I go about doing the username/password challenge through the socket without starting a new listening socket for mod_xml_curl requests .? I really appreciate any thoughts on this issue Many Thanks, From anthony.minessale at gmail.com Tue Mar 29 01:57:45 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Mar 2011 16:57:45 -0500 Subject: [Freeswitch-dev] mod_socket In-Reply-To: <293813.9658.qm@web120604.mail.ne1.yahoo.com> References: <293813.9658.qm@web120604.mail.ne1.yahoo.com> Message-ID: execute the "respond" application with "407" as the arg On Mon, Mar 28, 2011 at 4:45 PM, Ali R. wrote: > Hi everyone > I think my issue could be fixed with mod_xml_curl but I'm using FS purely > through sockets (inbound mode) > When I get the park event, I execute some biz logic.? Leg's A source IP plays a > role in this logic so based on the value of the IP (a big pool of allowed IP > addresses is generated dynamically and changes very fast so I have a logic to > query this pool) all good so far. However, if the incoming channel?s source IP > is not allowed I would still want to answer and continue on but I must challenge > it with a user name and password that I have a logic to retrieve.? Any thoughts > on how should I go about doing the username/password challenge through the > socket without starting a new listening socket for mod_xml_curl requests .? I > really appreciate any thoughts on this issue > > Many Thanks, > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From aroumie at yahoo.com Tue Mar 29 02:07:42 2011 From: aroumie at yahoo.com (Ali R.) Date: Mon, 28 Mar 2011 15:07:42 -0700 (PDT) Subject: [Freeswitch-dev] mod_socket In-Reply-To: References: <293813.9658.qm@web120604.mail.ne1.yahoo.com> Message-ID: <750883.12282.qm@web120615.mail.ne1.yahoo.com> Many thanks Anthony for the quick response; You are the hero. I will investigate and hope all goes well.... ----- Original Message ---- From: Anthony Minessale To: freeswitch-dev at lists.freeswitch.org Sent: Mon, March 28, 2011 2:57:45 PM Subject: Re: [Freeswitch-dev] mod_socket execute the "respond" application with "407" as the arg On Mon, Mar 28, 2011 at 4:45 PM, Ali R. wrote: > Hi everyone > I think my issue could be fixed with mod_xml_curl but I'm using FS purely > through sockets (inbound mode) > When I get the park event, I execute some biz logic.? Leg's A source IP plays a > role in this logic so based on the value of the IP (a big pool of allowed IP > addresses is generated dynamically and changes very fast so I have a logic to > query this pool) all good so far. However, if the incoming channel?s source IP > is not allowed I would still want to answer and continue on but I must >challenge > it with a user name and password that I have a logic to retrieve.? Any thoughts > on how should I go about doing the username/password challenge through the > socket without starting a new listening socket for mod_xml_curl requests .? I > really appreciate any thoughts on this issue > > Many Thanks, > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org From aroumie at yahoo.com Tue Mar 29 08:43:52 2011 From: aroumie at yahoo.com (Ali R.) Date: Mon, 28 Mar 2011 21:43:52 -0700 (PDT) Subject: [Freeswitch-dev] mod_socket In-Reply-To: <750883.12282.qm@web120615.mail.ne1.yahoo.com> References: <293813.9658.qm@web120604.mail.ne1.yahoo.com> <750883.12282.qm@web120615.mail.ne1.yahoo.com> Message-ID: <30510.52725.qm@web120616.mail.ne1.yahoo.com> Your suggestion coupled with chapter 10 in the book -page 258-260, I will be able to achieve 100% of my goal . With the "respond" app I can force the auth/challenge to take place based on biz logic (that's 50% of my problem fixed) and mod_xml_curl will hand the [user] back to me on a different socket in the form of HTTP request where I respond with the password from my biz logic. Now my question is, is there a way to negotiate the user/password through mod_socket. I'm avoiding to have my app listening for inbound connection to respond to curl?. While at it, FreeSwitch's slogan should be "FreeSwitch understands Security" I'm saying so because on page 260 of the FreeSwitch book, I found out that there is a way that you can pass back the password hashed instead of plain text which allow me to store it hashed in the database...small thing but really smart Regards, ----- Original Message ---- From: Ali R. To: freeswitch-dev at lists.freeswitch.org Sent: Mon, March 28, 2011 3:07:42 PM Subject: Re: [Freeswitch-dev] mod_socket Many thanks Anthony for the quick response; You are the hero. I will investigate and hope all goes well.... ----- Original Message ---- From: Anthony Minessale To: freeswitch-dev at lists.freeswitch.org Sent: Mon, March 28, 2011 2:57:45 PM Subject: Re: [Freeswitch-dev] mod_socket execute the "respond" application with "407" as the arg On Mon, Mar 28, 2011 at 4:45 PM, Ali R. wrote: > Hi everyone > I think my issue could be fixed with mod_xml_curl but I'm using FS purely > through sockets (inbound mode) > When I get the park event, I execute some biz logic. Leg's A source IP plays a > role in this logic so based on the value of the IP (a big pool of allowed IP > addresses is generated dynamically and changes very fast so I have a logic to > query this pool) all good so far. However, if the incoming channel?s source IP > is not allowed I would still want to answer and continue on but I must >challenge > it with a user name and password that I have a logic to retrieve. Any thoughts > on how should I go about doing the username/password challenge through the > socket without starting a new listening socket for mod_xml_curl requests . I > really appreciate any thoughts on this issue > > Many Thanks, > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org From anthony.minessale at gmail.com Tue Mar 29 18:50:54 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Mar 2011 09:50:54 -0500 Subject: [Freeswitch-dev] mod_socket In-Reply-To: <30510.52725.qm@web120616.mail.ne1.yahoo.com> References: <293813.9658.qm@web120604.mail.ne1.yahoo.com> <750883.12282.qm@web120615.mail.ne1.yahoo.com> <30510.52725.qm@web120616.mail.ne1.yahoo.com> Message-ID: You cannot manipulate the user directory over the socket since the lookup itself is triggered by xml lookups when the phone registers, so you would need to use xml_curl for this. You should probably just make a CGI and use a real web server to handle it. On Mon, Mar 28, 2011 at 11:43 PM, Ali R. wrote: > Your suggestion coupled with chapter 10 in the book -page 258-260, I will be > able to achieve 100% of my goal . ?With the "respond" app I can force the > auth/challenge to take place based on biz logic (that's 50% of my problem fixed) > and mod_xml_curl ?will hand the [user] back to me on a different socket in the > form of HTTP request where I respond with the password from my biz logic. Now my > question is, is there a way to negotiate the user/password through mod_socket. > I'm avoiding to have my app listening for inbound connection to respond to > curl?. ? While at it, FreeSwitch's slogan should be "FreeSwitch understands > Security" I'm saying so because on page 260 of the FreeSwitch book, I found out > that there is a way that you can pass back the password hashed instead of plain > text which allow me to store it hashed in the database...small thing but really > smart > > Regards, > > > > ----- Original Message ---- > From: Ali R. > To: freeswitch-dev at lists.freeswitch.org > Sent: Mon, March 28, 2011 3:07:42 PM > Subject: Re: [Freeswitch-dev] mod_socket > > Many thanks Anthony for the quick response; You are the hero. > I will investigate and hope all goes well.... > > > > ----- Original Message ---- > From: Anthony Minessale > To: freeswitch-dev at lists.freeswitch.org > Sent: Mon, March 28, 2011 2:57:45 PM > Subject: Re: [Freeswitch-dev] mod_socket > > execute the "respond" application with "407" as the arg > > > > > On Mon, Mar 28, 2011 at 4:45 PM, Ali R. wrote: >> Hi everyone >> I think my issue could be fixed with mod_xml_curl but I'm using FS purely >> through sockets (inbound mode) >> When I get the park event, I execute some biz logic. ?Leg's A source IP plays > a >> role in this logic so based on the value of the IP (a big pool of allowed IP >> addresses is generated dynamically and changes very fast so I have a logic to >> query this pool) all good so far. However, if the incoming channel?s source IP >> is not allowed I would still want to answer and continue on but I must >>challenge >> it with a user name and password that I have a logic to retrieve. ?Any > thoughts >> on how should I go about doing the username/password challenge through the >> socket without starting a new listening socket for mod_xml_curl requests . ?I >> really appreciate any thoughts on this issue >> >> Many Thanks, >> >> >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From aroumie at yahoo.com Tue Mar 29 20:44:03 2011 From: aroumie at yahoo.com (Ali R.) Date: Tue, 29 Mar 2011 09:44:03 -0700 (PDT) Subject: [Freeswitch-dev] mod_socket In-Reply-To: References: <293813.9658.qm@web120604.mail.ne1.yahoo.com> <750883.12282.qm@web120615.mail.ne1.yahoo.com> <30510.52725.qm@web120616.mail.ne1.yahoo.com> Message-ID: <853669.65043.qm@web120612.mail.ne1.yahoo.com> Thanks for taking the time to clarify it. A dedicated web server would scale better but this will be a big dependency. I will write a multi-threading module just to respond to xml_curl to emulate a web server and see how things perform with stress testing.... Thanks Again. ----- Original Message ---- From: Anthony Minessale To: freeswitch-dev at lists.freeswitch.org Sent: Tue, March 29, 2011 7:50:54 AM Subject: Re: [Freeswitch-dev] mod_socket You cannot manipulate the user directory over the socket since the lookup itself is triggered by xml lookups when the phone registers, so you would need to use xml_curl for this. You should probably just make a CGI and use a real web server to handle it. On Mon, Mar 28, 2011 at 11:43 PM, Ali R. wrote: > Your suggestion coupled with chapter 10 in the book -page 258-260, I will be > able to achieve 100% of my goal . With the "respond" app I can force the > auth/challenge to take place based on biz logic (that's 50% of my problem >fixed) > and mod_xml_curl will hand the [user] back to me on a different socket in the > form of HTTP request where I respond with the password from my biz logic. Now >my > question is, is there a way to negotiate the user/password through mod_socket. > I'm avoiding to have my app listening for inbound connection to respond to > curl?. While at it, FreeSwitch's slogan should be "FreeSwitch understands > Security" I'm saying so because on page 260 of the FreeSwitch book, I found out > that there is a way that you can pass back the password hashed instead of plain > text which allow me to store it hashed in the database...small thing but really > smart > > Regards, > > > > ----- Original Message ---- > From: Ali R. > To: freeswitch-dev at lists.freeswitch.org > Sent: Mon, March 28, 2011 3:07:42 PM > Subject: Re: [Freeswitch-dev] mod_socket > > Many thanks Anthony for the quick response; You are the hero. > I will investigate and hope all goes well.... > > > > ----- Original Message ---- > From: Anthony Minessale > To: freeswitch-dev at lists.freeswitch.org > Sent: Mon, March 28, 2011 2:57:45 PM > Subject: Re: [Freeswitch-dev] mod_socket > > execute the "respond" application with "407" as the arg > > > > > On Mon, Mar 28, 2011 at 4:45 PM, Ali R. wrote: >> Hi everyone >> I think my issue could be fixed with mod_xml_curl but I'm using FS purely >> through sockets (inbound mode) >> When I get the park event, I execute some biz logic. Leg's A source IP plays > a >> role in this logic so based on the value of the IP (a big pool of allowed IP >> addresses is generated dynamically and changes very fast so I have a logic to >> query this pool) all good so far. However, if the incoming channel?s source IP >> is not allowed I would still want to answer and continue on but I must >>challenge >> it with a user name and password that I have a logic to retrieve. Any > thoughts >> on how should I go about doing the username/password challenge through the >> socket without starting a new listening socket for mod_xml_curl requests . I >> really appreciate any thoughts on this issue >> >> Many Thanks, >> >> >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org From khovayko at gmail.com Wed Mar 30 04:44:55 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Tue, 29 Mar 2011 20:44:55 -0400 Subject: [Freeswitch-dev] Bruteforce hack Message-ID: <4D927D07.9010309@gmail.com> Hi, Couple months ago I have repotred strange behaviour of FreeSWITCH: 100% CPU usage, and leak memory. Today I catch this situation, and viewed logs. found something like ping-pong with SIP-port attack: FS log: 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.938408 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.957793 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.981472 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:42.999635 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.028769 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.048709 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.064379 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.080898 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.099860 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.118179 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.133097 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.149791 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.172722 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.187540 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.203845 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.219207 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.233950 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.250684 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.267531 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 2011-03-29 20:12:43.283970 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip 118.175.22.75 TCPDUMP output. 20:19:11.863940 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:19:11.914740 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 330 20:19:11.917521 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:19:11.931544 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 330 20:19:11.934351 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:19:11.946799 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:19:11.949370 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 330 20:19:11.954355 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:19:11.957996 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 20:19:11.958972 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 604 20:19:11.959998 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 20:19:11.961019 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 20:19:11.961814 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 329 20:19:11.962213 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 20:19:11.963141 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 20:19:11.964106 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 604 20:19:11.965059 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 604 20:19:11.966066 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 604 20:19:11.967018 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 604 20:19:11.967965 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 20:19:11.968930 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, length: 605 Also, I see, attack continues, when I stopped FreeSWITCH: 20:21:46.045995 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:21:46.046088 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.051280 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:21:46.051378 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.059059 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:21:46.059154 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.061089 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 330 20:21:46.061187 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.065982 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:21:46.066076 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.073622 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 330 20:21:46.073719 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.076260 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:21:46.076352 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.083160 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 331 20:21:46.083256 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 20:21:46.090586 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, length: 330 20:21:46.090684 IP deskpro.khovayko.com > 118.175.22.75: ICMP deskpro.khovayko.com udp port sip unreachable, length 36 So, you can see, this is not wrong FS-activity, this is just attack, attempt to hack in by method "brute force and ignorance". I think, easiest way to protect FS - to dynamically ban IP, from which comes attack. Or, maybe more smooth policy - to count attempts of unsuccessful login from some IP, and after threshold - set timewait for this IP. See following sample of pseudocode for demo this idea: char bantable[1<<12]; // 4K hashtable for ban counter. int ban_index = hash(user_ip_address) & (sizeof(bantable) - 1); if(user_login_success()) bantable[ban_index] = 0; // IP is valid else { if(bantable[ban_index] < 0) sleep(1); else bantable[ban_index]++; } Idea following - if real user comes in, and will be unlucky (hash as same as attacker), he just get 1s delay. But, for hacker, it will decrease attack dataflow, and slowing him... From jaybinks at gmail.com Wed Mar 30 04:52:38 2011 From: jaybinks at gmail.com (jay binks) Date: Wed, 30 Mar 2011 10:52:38 +1000 Subject: [Freeswitch-dev] Bruteforce hack In-Reply-To: <4D927D07.9010309@gmail.com> References: <4D927D07.9010309@gmail.com> Message-ID: check out Fail2Ban and turn on the correct sofia settings to enable fail2ban compatible logging. http://wiki.freeswitch.org/wiki/Fail2ban that will help protect you. I think the plan is not to bloat freeswitch with a million features, when they can be built around freeswitch.. also its better to do the banning at iptables that way the FS process dosnt even see the traffic. in my situation we take fail2ban a step further and we add a route to our BGP blackhole setup for repeat offenders. that way it dosnt even traverse the network. its just more flexible this way, then you can apply your own logic easily. Jay On Wed, Mar 30, 2011 at 10:44 AM, Oleg Khovayko wrote: > Hi, > > Couple months ago I have repotred strange behaviour of FreeSWITCH: 100% > CPU usage, > and leak memory. > > Today I catch this situation, and viewed logs. > found something like ping-pong with SIP-port attack: > > FS log: > > 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.754314 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.938408 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.957793 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.981472 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:42.999635 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.028769 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.048709 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.064379 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.080898 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.099860 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.118179 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.133097 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.149791 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.172722 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.187540 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.203845 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.219207 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.233950 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.250684 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.267531 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > 2011-03-29 20:12:43.283970 [WARNING] sofia_reg.c:1246 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [4 at 173.79.240.220] from ip > 118.175.22.75 > > TCPDUMP output. > > > 20:19:11.863940 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:19:11.914740 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 330 > 20:19:11.917521 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:19:11.931544 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 330 > 20:19:11.934351 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:19:11.946799 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:19:11.949370 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 330 > 20:19:11.954355 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:19:11.957996 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > 20:19:11.958972 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 604 > 20:19:11.959998 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > 20:19:11.961019 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > 20:19:11.961814 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 329 > 20:19:11.962213 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > 20:19:11.963141 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > 20:19:11.964106 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 604 > 20:19:11.965059 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 604 > 20:19:11.966066 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 604 > 20:19:11.967018 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 604 > 20:19:11.967965 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > 20:19:11.968930 IP deskpro.khovayko.com.sip > 118.175.22.75.5239: SIP, > length: 605 > > Also, I see, attack continues, when I stopped FreeSWITCH: > > 20:21:46.045995 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:21:46.046088 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.051280 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:21:46.051378 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.059059 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:21:46.059154 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.061089 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 330 > 20:21:46.061187 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.065982 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:21:46.066076 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.073622 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 330 > 20:21:46.073719 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.076260 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:21:46.076352 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.083160 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 331 > 20:21:46.083256 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > 20:21:46.090586 IP 118.175.22.75.5239 > deskpro.khovayko.com.sip: SIP, > length: 330 > 20:21:46.090684 IP deskpro.khovayko.com > 118.175.22.75: ICMP > deskpro.khovayko.com udp port sip unreachable, length 36 > > So, you can see, this is not wrong FS-activity, this is just attack, > attempt to hack in by method "brute force and ignorance". > > I think, easiest way to protect FS - to dynamically ban IP, from which > comes attack. > > Or, maybe more smooth policy - to count attempts of unsuccessful login > from some IP, and after threshold - set timewait for this IP. > > > See following sample of pseudocode for demo this idea: > > char bantable[1<<12]; // 4K hashtable for ban counter. > > int ban_index = hash(user_ip_address) & (sizeof(bantable) - 1); > > if(user_login_success()) > bantable[ban_index] = 0; // IP is valid > else { > if(bantable[ban_index] < 0) > sleep(1); > else > bantable[ban_index]++; > } > > Idea following - if real user comes in, and will be unlucky (hash as > same as attacker), he just get 1s delay. > But, for hacker, it will decrease attack dataflow, and slowing him... > > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110330/b023b5fc/attachment-0001.html From msc at freeswitch.org Wed Mar 30 05:49:03 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 Mar 2011 18:49:03 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Tomorrow - Author of SIPVicious Joining Us! Message-ID: Hello all! We have a special guest on our Wednesday conference call: Sandro Gauci, author of SIPVicious. He will be joining us to discuss SIPVicious and related issues. Please join us Wednesday, March 30th, at 1PM Eastern. Dial-in instructions can be found on the meeting agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_30 Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110329/81c6ccde/attachment.html From lautram.mathieu at gmail.com Wed Mar 30 20:20:12 2011 From: lautram.mathieu at gmail.com (Mathieu Lautram) Date: Wed, 30 Mar 2011 18:20:12 +0200 Subject: [Freeswitch-dev] stuck channels Message-ID: Hi everyone, I have a problem of stuck channels. Indeed, when I do a show channels in fs_cli, there is a lot of hangup callstate channels. I even tried to do a uuid_kill but fs_cli tells me "-ERR No Such Channels"... So I d'ont know where the problem is. Someone tells me that it could be a sqlite problem. I hope you could help me because it's been 3 weeks that I'm on it and I have no solutions... Here is the log corresponding to my problem: 2011-03-30 15:26:54.866347 [NOTICE] sofia.c:537 Hangup sofia/external/ 0142560840 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:26:54.866347 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0142560840 at 192.168.0.1 [KILL] 2011-03-30 15:26:54.866347 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0142560840 at 192.168.0.1 [BREAK] 2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2563 (sofia/external/ 0134747924 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:26:56.268289 [NOTICE] sofia.c:537 Hangup sofia/external/ 0134747924 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:26:56.268289 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0134747924 at 192.168.0.1 [KILL] 2011-03-30 15:26:56.268289 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0134747924 at 192.168.0.1 [BREAK] 2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2563 (sofia/external/ 0139339445 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:00.066249 [NOTICE] sofia.c:537 Hangup sofia/external/ 0139339445 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:00.066249 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0139339445 at 192.168.0.1 [KILL] 2011-03-30 15:27:00.066249 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0139339445 at 192.168.0.1 [BREAK] 2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2563 (sofia/external/ 0143708446 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:00.563121 [NOTICE] sofia.c:537 Hangup sofia/external/ 0143708446 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:00.563121 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0143708446 at 192.168.0.1 [KILL] 2011-03-30 15:27:00.563121 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0143708446 at 192.168.0.1 [BREAK] 2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2563 (sofia/external/ 0148863352 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:01.376088 [NOTICE] sofia.c:537 Hangup sofia/external/ 0148863352 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:01.376088 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0148863352 at 192.168.0.1 [KILL] 2011-03-30 15:27:01.376088 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0148863352 at 192.168.0.1 [BREAK] 2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2563 (sofia/external/ 0140411621 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:01.377142 [NOTICE] sofia.c:537 Hangup sofia/external/ 0140411621 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:01.377142 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0140411621 at 192.168.0.1 [KILL] 2011-03-30 15:27:01.377142 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0140411621 at 192.168.0.1 [BREAK] 2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2563 (sofia/external/ 0130510861 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:04.264971 [NOTICE] sofia.c:537 Hangup sofia/external/ 0130510861 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:04.264971 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0130510861 at 192.168.0.1 [KILL] 2011-03-30 15:27:04.264971 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0130510861 at 192.168.0.1 [BREAK] 2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2563 (sofia/external/ 0134868273 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:05.165960 [NOTICE] sofia.c:537 Hangup sofia/external/ 0134868273 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:05.165960 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0134868273 at 192.168.0.1 [KILL] 2011-03-30 15:27:05.165960 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0134868273 at 192.168.0.1 [BREAK] 2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2563 (sofia/external/ 0142861137 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:05.665936 [NOTICE] sofia.c:537 Hangup sofia/external/ 0142861137 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:05.665936 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0142861137 at 192.168.0.1 [KILL] 2011-03-30 15:27:05.665936 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0142861137 at 192.168.0.1 [BREAK] 2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2563 (sofia/external/ 0139880615 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:05.962900 [NOTICE] sofia.c:537 Hangup sofia/external/ 0139880615 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:05.962900 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0139880615 at 192.168.0.1 [KILL] 2011-03-30 15:27:05.962900 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0139880615 at 192.168.0.1 [BREAK] 2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2563 (sofia/external/ 0140209135 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:08.964780 [NOTICE] sofia.c:537 Hangup sofia/external/ 0140209135 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:08.964780 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0140209135 at 192.168.0.1 [KILL] 2011-03-30 15:27:08.964780 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0140209135 at 192.168.0.1 [BREAK] 2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2563 (sofia/external/ 0134524074 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:09.664751 [NOTICE] sofia.c:537 Hangup sofia/external/ 0134524074 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:09.664751 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0134524074 at 192.168.0.1 [KILL] 2011-03-30 15:27:09.664751 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0134524074 at 192.168.0.1 [BREAK] 2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2563 (sofia/external/ 0130326184 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:09.767745 [NOTICE] sofia.c:537 Hangup sofia/external/ 0130326184 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:09.767745 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0130326184 at 192.168.0.1 [KILL] 2011-03-30 15:27:09.767745 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0130326184 at 192.168.0.1 [BREAK] 2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2563 (sofia/external/ 0139902937 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:09.864742 [NOTICE] sofia.c:537 Hangup sofia/external/ 0139902937 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:09.864742 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0139902937 at 192.168.0.1 [KILL] 2011-03-30 15:27:09.864742 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0139902937 at 192.168.0.1 [BREAK] 2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2563 (sofia/external/ 0144311010 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:12.064653 [NOTICE] sofia.c:537 Hangup sofia/external/ 0144311010 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:12.064653 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0144311010 at 192.168.0.1 [KILL] 2011-03-30 15:27:12.064653 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0144311010 at 192.168.0.1 [BREAK] 2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2563 (sofia/external/ 0139199893 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:12.466637 [NOTICE] sofia.c:537 Hangup sofia/external/ 0139199893 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:12.466637 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0139199893 at 192.168.0.1 [KILL] 2011-03-30 15:27:12.466637 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0139199893 at 192.168.0.1 [BREAK] 2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2563 (sofia/external/ 0142533724 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:14.062573 [NOTICE] sofia.c:537 Hangup sofia/external/ 0142533724 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:14.062573 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0142533724 at 192.168.0.1 [KILL] 2011-03-30 15:27:14.062573 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0142533724 at 192.168.0.1 [BREAK] 2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2563 (sofia/external/ 0141322567 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:14.066573 [NOTICE] sofia.c:537 Hangup sofia/external/ 0141322567 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:14.066573 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0141322567 at 192.168.0.1 [KILL] 2011-03-30 15:27:14.066573 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0141322567 at 192.168.0.1 [BREAK] 2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2563 (sofia/external/ 0130344121 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:16.564476 [NOTICE] sofia.c:537 Hangup sofia/external/ 0130344121 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:16.564476 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0130344121 at 192.168.0.1 [KILL] 2011-03-30 15:27:16.564476 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0130344121 at 192.168.0.1 [BREAK] 2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2563 (sofia/external/ 0139595901 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:26.464071 [NOTICE] sofia.c:537 Hangup sofia/external/ 0139595901 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:26.464071 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0139595901 at 192.168.0.1 [KILL] 2011-03-30 15:27:26.464071 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0139595901 at 192.168.0.1 [BREAK] 2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2563 (sofia/external/ 0143758609 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:27.066047 [NOTICE] sofia.c:537 Hangup sofia/external/ 0143758609 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:27.066047 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0143758609 at 192.168.0.1 [KILL] 2011-03-30 15:27:27.066047 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0143758609 at 192.168.0.1 [BREAK] 2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2563 (sofia/external/ 0148831626 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:29.953931 [NOTICE] sofia.c:537 Hangup sofia/external/ 0148831626 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:29.953931 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0148831626 at 192.168.0.1 [KILL] 2011-03-30 15:27:29.953931 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0148831626 at 192.168.0.1 [BREAK] 2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2563 (sofia/external/ 0140419154 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:36.965646 [NOTICE] sofia.c:537 Hangup sofia/external/ 0140419154 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:36.965646 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0140419154 at 192.168.0.1 [KILL] 2011-03-30 15:27:36.965646 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0140419154 at 192.168.0.1 [BREAK] 2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2563 (sofia/external/ 0142569040 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:53.565145 [NOTICE] sofia.c:537 Hangup sofia/external/ 0142569040 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:53.565145 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0142569040 at 192.168.0.1 [KILL] 2011-03-30 15:27:53.565145 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0142569040 at 192.168.0.1 [BREAK] 2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2563 (sofia/external/ 0140261579 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:27:59.260817 [NOTICE] sofia.c:537 Hangup sofia/external/ 0140261579 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:27:59.260817 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0140261579 at 192.168.0.1 [KILL] 2011-03-30 15:27:59.260817 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0140261579 at 192.168.0.1 [BREAK] 2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2563 (sofia/external/ 0134291660 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:28:07.062534 [NOTICE] sofia.c:537 Hangup sofia/external/ 0134291660 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:28:07.062534 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0134291660 at 192.168.0.1 [KILL] 2011-03-30 15:28:07.062534 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0134291660 at 192.168.0.1 [BREAK] 2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2563 (sofia/external/ 0142335551 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:28:08.761352 [NOTICE] sofia.c:537 Hangup sofia/external/ 0142335551 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:28:08.761352 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0142335551 at 192.168.0.1 [KILL] 2011-03-30 15:28:08.761352 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0142335551 at 192.168.0.1 [BREAK] 2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2563 (sofia/external/ 0145932250 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:28:26.160649 [NOTICE] sofia.c:537 Hangup sofia/external/ 0145932250 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:28:26.160649 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0145932250 at 192.168.0.1 [KILL] 2011-03-30 15:28:26.160649 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0145932250 at 192.168.0.1 [BREAK] 2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2563 (sofia/external/ 0146377247 at 192.168.0.1) Callstate Change ACTIVE -> HANGUP 2011-03-30 15:28:31.958419 [NOTICE] sofia.c:537 Hangup sofia/external/ 0146377247 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2011-03-30 15:28:31.958419 [DEBUG] switch_channel.c:2579 Send signal sofia/external/0146377247 at 192.168.0.1 [KILL] 2011-03-30 15:28:31.958419 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/0146377247 at 192.168.0.1 [BREAK] freeswitch at internal> freeswitch at internal> freeswitch at internal> show channels 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 7c91effe-7af7-47cc-a9c2-5a8d7a13e070,outbound,2011-03-30 15:24:18,1301491458,sofia/external/0146426664 at 192.168.0.1 ,CS_EXECUTE,,0000000000,,0146426664,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0146426664,0146426664,RECV,d2ea58aa-e3a4-496e-a031-ac501910d18e 19b44e66-0cc8-4bf2-b8e4-8587986c4e22,outbound,2011-03-30 15:24:46,1301491486,sofia/external/0140675411 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140675411,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0140675411,RECV,472fb609-8d90-4d7d-bc0f-7353bb01139f 4dca35aa-589b-4ba8-9efa-f7510514385b,outbound,2011-03-30 15:25:02,1301491502,sofia/external/0134291660 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134291660,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0134291660,RECV,f62c2f5d-0b17-4e1e-8f5c-9c495407e95e 309fd56c-9d4b-4904-be94-8803cc757dd3,outbound,2011-03-30 15:25:08,1301491508,sofia/external/0146377247 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146377247,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0146377247,RECV,cdcfd0eb-5efc-4541-9abd-845abfa63fa9 384fdd12-908c-4ad5-a488-a67f25b253ed,outbound,2011-03-30 15:25:20,1301491520,sofia/external/0141955961 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141955961,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0141955961,RECV,f1a4da83-7ff9-478b-9323-0b9a69104ea2 66ba4d1c-a8de-4a5f-8da3-53eea2e353be,outbound,2011-03-30 15:25:22,1301491522,sofia/external/0146770127 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146770127,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0146770127,RECV,50a004d4-bb3e-4baf-afe5-a9eeb5a58cfc 398d2132-7f7a-4ddf-95af-fe8afcb2a253,outbound,2011-03-30 15:25:23,1301491523,sofia/external/0144598175 at 192.168.0.1,CS_EXECUTE,,0000000000,,0144598175,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0144598175,RECV,98170d54-5c3d-40be-8d20-30da9a478730 2b90e1ba-16c4-4d48-bc4f-90c753a9244e,outbound,2011-03-30 15:25:23,1301491523,sofia/external/0130310549 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130310549,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0130310549,RECV,5a3cba1e-f559-4116-93d8-3cca45e3084b 13671770-a92a-4f18-98bb-e8488d926cdb,outbound,2011-03-30 15:25:25,1301491525,sofia/external/0148863352 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148863352,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0148863352,RECV,0c01dc58-90ed-408f-a574-1091c35b2c13 6da9d0de-ef0d-4dfc-85bf-d771b3c6f098,outbound,2011-03-30 15:25:25,1301491525,sofia/external/0146682550 at 192.168.0.1,CS_EXECUTE,,0000000000,,0146682550,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0146682550,RECV,e8754f05-8870-4b25-8f7a-107ee129b1e2 cb332c48-443b-4326-86b4-ea3d89346fef,outbound,2011-03-30 15:25:25,1301491525,sofia/external/0142378320 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142378320,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0142378320,RECV,0759c1a2-9f3a-4259-9e3a-7f567318cf67 4f960d60-8079-4225-bea2-58f8ccba67f5,outbound,2011-03-30 15:25:26,1301491526,sofia/external/0148521177 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148521177,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0148521177,RECV,37b401ba-3077-43b8-bf8a-ee3a836ac843 c03c2320-1ddb-40b0-884b-d4b07ac4d189,outbound,2011-03-30 15:25:27,1301491527,sofia/external/0142569040 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142569040,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0142569040,RECV,1be9a468-b720-4398-b8c6-621c52aad6ac b4a6ef35-79d3-4520-9574-1c37c2fecec7,outbound,2011-03-30 15:25:28,1301491528,sofia/external/0140411621 at 192.168.0.1 ,CS_EXECUTE,,0000000000,,0140411621,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0140411621,0140411621,RECV,81e14c30-0b77-46e6-96a4-ae8a0678e6c5 67b126e7-2a5f-4b99-9843-ee71e2c08c10,outbound,2011-03-30 15:25:35,1301491535,sofia/external/0139745628 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139745628,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0139745628,RECV,1b173b0d-69c6-40e0-8b71-1496106cbead dbd23a86-64bf-417d-bb77-6b4a927b9efd,outbound,2011-03-30 15:25:35,1301491535,sofia/external/0139595901 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139595901,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0139595901,RECV,395e8762-4f70-48f3-b8b5-b5040a2e5fbe dfe644c4-d453-4c39-93bd-34f7f3e9b570,outbound,2011-03-30 15:25:36,1301491536,sofia/external/0142335551 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142335551,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0142335551,RECV,51995424-c0f4-4252-b4ad-3b4ae7fcdad6 96a82658-74eb-4508-8f6b-7d88705274cb,outbound,2011-03-30 15:25:36,1301491536,sofia/external/0147662091 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147662091,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0147662091,RECV,2a5464d8-c50d-4088-bfaa-0e23ff24db08 9d507251-73f6-4f96-9026-37eeabae525c,outbound,2011-03-30 15:25:38,1301491538,sofia/external/0147050416 at 192.168.0.1,CS_EXECUTE,,0000000000,,0147050416,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0147050416,RECV,59603591-4298-41f8-af0a-ad939e0b16f2 0a327bf4-b3a1-43f1-bcfa-e8c8770f8a2b,outbound,2011-03-30 15:25:39,1301491539,sofia/external/0143387777 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143387777,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0143387777,RECV,dc9713a8-57a9-465c-bd6e-006aadb67903 c769b907-cea2-41c4-a3a6-1b1699960f27,outbound,2011-03-30 15:25:39,1301491539,sofia/external/0143758609 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143758609,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0143758609,RECV,2c9b1ab3-d1ee-499b-8771-91168acbd091 99cabd94-db28-47cb-8257-c94be0cd279f,outbound,2011-03-30 15:25:40,1301491540,sofia/external/0134162490 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134162490,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0134162490,RECV,3ed6414e-e3e7-4176-986d-47f69f6dd2c4 89b66ced-eea9-4f0c-a56c-7b8a96a0416b,outbound,2011-03-30 15:25:42,1301491542,sofia/external/0145932250 at 192.168.0.1,CS_EXECUTE,,0000000000,,0145932250,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0145932250,RECV,f420c519-b869-48fc-8190-4cf0e86c36e3 5a1a30c4-5df7-4618-b52c-ab3e4782734c,outbound,2011-03-30 15:25:44,1301491544,sofia/external/0143318179 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143318179,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0143318179,RECV,84e3bb19-a94f-4d26-ba46-5a3347644377 d07e50bd-303f-4181-85f2-27bdbb74b93d,outbound,2011-03-30 15:25:45,1301491545,sofia/external/0143708446 at 192.168.0.1,CS_EXECUTE,,0000000000,,0143708446,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0143708446,RECV,70c6ec80-78fe-4f1f-9e3a-732d544f74f2 dab44fd9-8f4b-42fd-855e-9ab4a5723687,outbound,2011-03-30 15:25:46,1301491546,sofia/external/0134988997 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134988997,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0134988997,RECV,2b4c5913-4131-4387-94d1-30ab4366a49e 0ec7264f-5eaf-4f19-9b40-792ccf62b401,outbound,2011-03-30 15:25:47,1301491547,sofia/external/0141322567 at 192.168.0.1,CS_EXECUTE,,0000000000,,0141322567,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0141322567,RECV,15d192af-716e-44cb-9d00-08a9ebbdbb79 5a984ed3-6dd8-4991-912d-877469896bfa,outbound,2011-03-30 15:25:49,1301491549,sofia/external/0145234010 at 192.168.0.1,CS_EXECUTE,,0000000000,,0145234010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,Outbound Call,0145234010,RECV,ee680d98-9bec-4830-ae67-9d8a70aa3b08 688a2a96-bb50-4d12-8fe9-ec9423a42bd7,outbound,2011-03-30 15:25:49,1301491549,sofia/external/0134747924 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134747924,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0134747924,RECV,fb9aecb6-85e8-4d3e-8c0c-b8a252425e71 3402d9bd-e33e-4e6f-9bc2-43ab559fa8aa,outbound,2011-03-30 15:25:49,1301491549,sofia/external/0147239297 at 192.168.0.1 ,CS_EXECUTE,,0000000000,,0147239297,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,ACTIVE,0147239297,0147239297,RECV,0924c5b1-fd35-40de-b47e-526b767cfce1 66101a12-5c62-4d03-9eb3-dd0a7a73a22d,outbound,2011-03-30 15:25:50,1301491550,sofia/external/0142533724 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142533724,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0142533724,RECV,55115eca-c7c3-4657-98a8-fc21f13a10c9 905d5a84-c7e2-4218-881f-39655efd040a,outbound,2011-03-30 15:25:50,1301491550,sofia/external/0140261579 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140261579,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0140261579,RECV,b79dc96f-146d-4a94-9025-ea82e06d30e5 0cc72f20-e347-47ad-aa7a-73cf5e486770,outbound,2011-03-30 15:25:53,1301491553,sofia/external/0141734848 at 192.168.0.1 ,CS_EXECUTE,,0000000000,,0141734848,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,0141734848,0141734848,RECV,f3f48827-fe5c-4b92-a83e-8f1e88f5263d 6a7c903d-a1b8-4898-92a9-17e8e21bc477,outbound,2011-03-30 15:25:53,1301491553,sofia/external/0130926822 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130926822,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0130926822,RECV,1a6d58ae-d93b-4a82-a311-e478dcf6c972 cf48337a-bd0e-4739-8aad-dd01534de325,outbound,2011-03-30 15:25:55,1301491555,sofia/external/0139194547 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139194547,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0139194547,RECV,2687f485-f872-44e9-b04b-3ab45ccdbda2 9c4e2841-0d70-49bd-9876-298cdbfd07d8,outbound,2011-03-30 15:26:01,1301491561,sofia/external/0142560840 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142560840,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0142560840,RECV,223d8889-5006-4043-8792-88f0059a726f 097527a5-af50-4544-95f9-41e5d2db4329,outbound,2011-03-30 15:26:01,1301491561,sofia/external/0139339445 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139339445,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0139339445,RECV,ef818d92-7a22-4d00-8e17-8f6428db7255 0c734637-b2fb-4b3d-b4f3-33bc1d6565e5,outbound,2011-03-30 15:26:05,1301491565,sofia/external/0134868273 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134868273,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0134868273,RECV,2364ea0b-d80c-44d5-915c-b4ee0498f50d 65a8d0fc-26e9-4f21-9117-1d833111a428,outbound,2011-03-30 15:26:07,1301491567,sofia/external/0139199893 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139199893,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0139199893,RECV,a14ad031-c059-45a7-a822-4fbfddeb56fd ea33d5ab-4dc5-45a8-8e5d-6dfbeb65c1b0,outbound,2011-03-30 15:26:08,1301491568,sofia/external/0140209135 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140209135,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0140209135,RECV,0786ebcf-be37-48f0-b00c-6e1b6f22af75 473be842-169d-4ea3-8c86-c7f1d0bc01db,outbound,2011-03-30 15:26:08,1301491568,sofia/external/0142861137 at 192.168.0.1,CS_EXECUTE,,0000000000,,0142861137,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0142861137,RECV,b4ae8669-20f0-423f-951a-a14affbe76e1 f25d32b4-3b8e-42aa-a38a-b3fc7b11bf97,outbound,2011-03-30 15:26:09,1301491569,sofia/external/0134524074 at 192.168.0.1,CS_EXECUTE,,0000000000,,0134524074,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0134524074,RECV,9f82e9cb-5f79-4f76-8e1e-7f8cf8fb39e1 0694de2a-1251-4ba7-a5e3-302403bda1ec,outbound,2011-03-30 15:26:09,1301491569,sofia/external/0130326184 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130326184,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0130326184,RECV,3dd6da1f-a59d-4b72-8b87-0657aee2a7eb 6db28a24-99c7-4050-9c7e-7df94ee05860,outbound,2011-03-30 15:26:10,1301491570,sofia/external/0130510861 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130510861,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0130510861,RECV,f1d0b27f-a7d0-4dd0-a2a1-873525e6ed47 676d1965-4bd0-43a3-85a2-84d1c618098c,outbound,2011-03-30 15:26:12,1301491572,sofia/external/0139880615 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139880615,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0139880615,RECV,69fb0f7e-b279-46a0-a9ff-0a685b2a2d9a 10efef01-29bb-4181-bf34-70390cdbe964,outbound,2011-03-30 15:26:13,1301491573,sofia/external/0139902937 at 192.168.0.1,CS_EXECUTE,,0000000000,,0139902937,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0139902937,RECV,505c212f-7459-4ace-836c-7d35737a5749 9efbd4c6-17f9-4035-b185-84c53f8e67f5,outbound,2011-03-30 15:26:14,1301491574,sofia/external/0130344121 at 192.168.0.1,CS_EXECUTE,,0000000000,,0130344121,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0130344121,RECV,2a79f551-dd7a-454d-af0b-b893a05a3bee f111d343-9a56-4d18-890f-9482f7743264,outbound,2011-03-30 15:26:16,1301491576,sofia/external/0144311010 at 192.168.0.1,CS_EXECUTE,,0000000000,,0144311010,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0144311010,RECV,eb7e7eed-d4d5-46ad-ac91-6a90b3bab9b6 a2f60b85-204b-4dd7-933d-af04d5807b3c,outbound,2011-03-30 15:26:32,1301491592,sofia/external/0148831626 at 192.168.0.1,CS_EXECUTE,,0000000000,,0148831626,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0148831626,RECV,3bd290c2-2275-428c-9edd-e940c659e730 9a40f095-69e6-4a1b-97ea-d464c6ea2f4a,outbound,2011-03-30 15:26:39,1301491599,sofia/external/0140419154 at 192.168.0.1,CS_EXECUTE,,0000000000,,0140419154,txfax,/var/www/html/faxbroadcast_free/log/1301482273-80.13.58.1--fax-change-ton-numero-15-02-2011_BULLZIP.tiff,,default,L16,8000,128000,PCMU,8000,64000,,localhost.localdomain,,,HANGUP,Outbound Call,0140419154,RECV,d79fa0b6-6e77-4fb8-8d37-bcd1d7f68a8d 50 total. Thank you =) -- Mathieu LAUTRAM Application developer BJT Partners - FRANCE +33 1 79 75 99 60 +33 6 61 59 07 25 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110330/a1f9c5b3/attachment-0001.html From jpatten at co.brazos.tx.us Wed Mar 30 18:08:45 2011 From: jpatten at co.brazos.tx.us (Josh M. Patten) Date: Wed, 30 Mar 2011 14:08:45 +0000 Subject: [Freeswitch-dev] Valet Parking improvements Message-ID: <8C8A3D4965236A42BDFF1758727F049A263BAD@ITEX1.bc.local> Our development team is in the process of writing some improvements into the valet parking module that we hope to get pushed upstream to the community. Among these improvements are: * Ability to set timeout value o Ability to send call back to extension that originally parked the call after timeout o Ability to send call to static extension (for example, an operator) after timeout o Ability to disconnect the call after timeout * Ability to set escape button, such as *, #, 0-9 o Ability to send call back to extension that originally parked the call after caller presses escape button o Ability to send call to static extension (for example, an operator) after caller presses escape button o Ability to disconnect the call after caller presses escape button In order to make these changes we are going to have to add a few arguments to the module. My questions are: How should we get this code submitted for review and testing? What do we need to do to get this code merged after it is reviewed and tested? Thanks! Josh Patten Brazos County Network Engineer 979.361.4676 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110330/9a50233b/attachment.html From andrew.keil at askinteractive.net Tue Mar 29 06:10:56 2011 From: andrew.keil at askinteractive.net (Andrew Keil) Date: Tue, 29 Mar 2011 13:10:56 +1100 Subject: [Freeswitch-dev] Freeswitch latest source rebuild issue under Windows Message-ID: To Freeswitch developers.... OK. Here goes, what I have done so far... 1) Setup a Virtual Machine running 32-bit XP SP3 (with all Critical Updates applied). That way I can quickly go back in time and try an install again if necessary. 2) Since I noticed that you now support Visual C++ 2010 Express, I downloaded and installed that (without the option for SQL Server Express 2008). 3) I also then installed the Windows SDK (7.1 (the latest version)). The reason for this is when I download your latest build of Freeswitch 1.0.7 today it complained about 64bit settings throughout various project files when opening Freeswitch.2010.express.sln 4) Then I simply downloaded your latest build from http://latest.freeswitch.org/ and extracted it out to c:\FreeSWITCH (so that there is a sub-directory freeswitch-1.0.7 below it with all the files) - this looks the same layout as the Freeswitch book example, except 1.0.7 version instead. 5) Opened the Freeswitch.2010.express.sln 5.1) Stated inside the Visual Studio Output Window: "Some of the properties associated with the solution could not be read." 6) Built the solution via F7 See attached for the build errors from Visual Studio. I guess my questions are the following: Q1) Should I be running Visual C++ 2010 Express or Visual C++ 2008 Express - it does not matter to me, however I always like to move with the times? Q2) Was I right in installing the Windows SDK (http://msdn.microsoft.com/en-us/windows/bb980924.aspx) after the installation of Visual C++ 2010 Express? Perhaps this should be added to the Freeswitch windows installation instructions page: http://wiki.freeswitch.org/wiki/Installation_for_Windows Q3) Should I expect the http://latest.freeswitch.org/ version of freeswitch to Build first time? Has this been tested? Q4) Should I simply use the 1.0.6 version with Visual C++ 2008 Express and follow the Freeswitch book word for word? This seems a reliable path, but I am a 'C' developer myself and would like to start out with a more up-to-date dev. environment and a version of freeswitch that is closer to the current version. Q5) Can someone make comments on the Build errors that are inside the attached file. Obviously I could go through and debug each build error, however I feel it best to have my questions answered above to avoid wasting time. Please note. I can happily start again and go back in time to the point prior to the installation of Visual C++ 2010 Express, since I am running a ESXi Virtual Machine. All I aiming for is a clean setup of FreeSWITCH under XP for testing and developing purposes (this is not to run in a production environment). I would like to be as up-to-date as possible in order to progress quickly (obviously I will install GIT/TortoiseGIT etc.. to make my dev. environment more integrated once I have a working base version of Freeswitch). Thanks in advance, Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110329/0eac40fd/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: Freeswitch-1.0.7_VC_2010_Express_InitialBuildErrors.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110329/0eac40fd/attachment-0001.txt From barisyanar at gmail.com Thu Mar 24 18:43:42 2011 From: barisyanar at gmail.com (barisyanar) Date: Thu, 24 Mar 2011 15:43:42 -0000 Subject: [Freeswitch-dev] module load problem Message-ID: Hi, I compiled a module using intel ipp library and freeswitch 1.0.7 source. SWITCH_API_VERSION is 5 in switch_types.h However I cannot load the module with fs_cli. Below is the error I get: 2011-03-24 16:35:00.584545 [CRIT] switch_loadable_module.c:928 Error Loading module /opt/freeswitch/mod/mod_tri.so **Trying to load an out of date module, please rebuild the module.** How could this happen? I expect the .so file has the same API_VERSION with the freeswitch. Baris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110324/46bb4156/attachment.html From casanova at brastel.co.jp Thu Mar 24 14:43:15 2011 From: casanova at brastel.co.jp (Daniel Casanova) Date: Thu, 24 Mar 2011 11:43:15 -0000 Subject: [Freeswitch-dev] Issue with in-band DTMF when using session:recordFile In-Reply-To: References: Message-ID: 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 From rhuddleston at gmail.com Thu Mar 24 20:29:45 2011 From: rhuddleston at gmail.com (Robert Huddleston) Date: Thu, 24 Mar 2011 17:29:45 -0000 Subject: [Freeswitch-dev] CoreDump - Possibly User Error Message-ID: <1ec301cbea49$19c4a120$4d4de360$@com> http://pastebin.freeswitch.org/15833 I was editing mod_xml_cdr config file while FreeSwitch was up and running. Tried to reconnect to fs_cli and could not connect. Found that FreeSwitch had coredumped. Tried to use GDB BT but it's complaining about some symbols missing - which is odd as I built the box from GIT and source is installed. I admit I'm a n00b when it comes to GDB and BT Any help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110324/843fbc79/attachment.html From ritzalam at gmail.com Wed Mar 23 00:47:05 2011 From: ritzalam at gmail.com (Richard Alam) Date: Tue, 22 Mar 2011 21:47:05 -0000 Subject: [Freeswitch-dev] Questions on implementing mod_nelly codec In-Reply-To: References: Message-ID: Yes, we plan on contributing the module once it's working. On Tue, Mar 22, 2011 at 5:19 PM, Anthony Minessale wrote: > Are you planning to contribute this module? > > On Mon, Mar 21, 2011 at 1:58 PM, Richard Alam wrote: >> Hi, >> >> We've been trying to implement a mod_nelly codec but so far has been >> unsuccessful. We use a Nellymoser compatible codec submitted to ffmpeg >> (see June 16, 2008 entry of http://ffmpeg.org/). >> >> Our plan is to use Flash using Nellymoser to connect to a conference >> in FreeSWITCH. >> >> The hurdle seems to be that Nellymoser decodes 64 byte nelly audio to >> 512 byte L16. Nellymoser is 11-Khz 8-bit per sample with 32 ptime. >> However, L16 implementations are all 20 ptime >> which have uncompressed bytes in multiples of 320. This doesn't align >> with what mod_nelly expects when encoding (convert 512 bytes to 64 >> bytes) and decoding (64 bytes to 512 bytes). >> >> Here is the SWITCH_MODULE_LOAD_FUNCTION(mod_nelly_load) >> [http://pastebin.freeswitch.org/15764] now. We've played around with >> the samples per second (SPS), actual samples per second (ASPS), >> bits per second (BPS), and the ptime (PTIME) but with no luck. It's >> either we get bad very choppy audio, incompatible destination (when >> codecs can't match), or no audio at all. We've managed to get the >> "You will now be placed into the conference" audio played correctly >> but the audio from the caller is bad. Just the paying of wav file from >> FS is correct. >> >> My question is how does transcoding works in FS? Am I correct to >> assume decoding as Nelly->L16 and encoding as L16->Nelly? >> So basically the media flow for our use case is Nelly -> L16 --> >> mod_conference -> L16 -> Nelly. >> >> Do I need to make the frames align? In this case, Nelly expect 256 >> samples per frame while L16 has 160, 320, ... samples per frame. >> >> Hope I've articulated clearly what we are trying to accomplish. >> >> Suggestions and ideas on what to try out next is much appreciated. >> >> Thanks in advance. >> >> Richard >> >> >> -- >> --- >> BigBlueButton >> http://www.bigbluebutton.org >> http://code.google.com/p/bigbluebutton >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton