December 2008 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

mod_skypiax inching forward
by Giovanni Maruzzelli 09 Jan '09

09 Jan '09

Hi FreeSWITCHers! mod_skypiax, the Skype compatible endpoint, is slowly inching toward release :-) When the demo is online (will go on and off for development), you can test it (so helping finding bugs) by calling with Skype the Skype Names: skypiax20, skypiax19, skypiax18, ...., skypiax1 Happy New Year !!! Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039

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Audio formats without transcoding
by Juan Jose Comellas 02 Jan '09

02 Jan '09

I need to support calls using G.711, G.729 and G.722 and I want to avoid transcoding as much as possible. Does FreeSWITCH support playing audios in a raw format that does not require transcoding for any of these codecs? It looks like FS provides WAV audio files with the frequencies of the supported codecs and relies on transcoding all the time. Is my analysis correct? How does everybody else handle this problem? Thanks.

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FreeSWITCH News: New FreeSWITCH Client Available!
by Michael Collins 30 Dec '08

30 Dec '08

Hello FreeSWITCHers! Just a heads up, there are lots of cool things happening with FreeSWITCH. Please check out the latest here: http://freeswitch.org/node/156 Stay tuned for more news from the FreeSWITCH camp. -MC (mercutioviz)

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Re: [Freeswitch-dev] Build on VC++ 2008 Express
by Yours 28 Dec '08

28 Dec '08

Dear all friends, I understand some problems with VC 2008 express. When I run build,it always connect to svn to get some package automatically, last time, I run VC without internet, so it was failed. today, I suceeded in compile freeswitch after I build three times, in spite there are 3 error messages like following.I can run the freeswitch.exe from debug folder. Thank all friend in the project. the error message are: msg_parser.c ..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c : error C2220: no object is generated ..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c : warning C4819: The file contains a character that can ot be represented in the current code page(936). save the file in unicode format to prevent data loss ..\..\sofia-sip\libsofia-sip-ua\nua\nua.c : error C2220: no object is generated ..\..\sofia-sip\libsofia-sip-ua\nua\nua.c : warning C4819: The file contains a character that can ot be represented in the current code page(936). save the file in unicode format to prevent data loss LINK : fatal error LINK1181: cannot open input file “..\..\..\..\libs\win32\sofia\debug\libsofia_sip_ua_static.lib” sincerely London Hood

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Implicit Recording Pause
by Robert Clayton 27 Dec '08

27 Dec '08

All, I thought I would try again. This question has been pending for the past 22 days on both the dev and users forums. If I am not understanding the process for asking questions please inform me. I was thinking since there is no direct functionality for the person being recorded to pause the recording could this be done indirectly. For example when using the functionality to record only when voice is present could the audio stream be preprocessed to evaluate the dtmf where if a specific key was pressed the audio stream would no longer be passed or passed as blank to the recording functionality so an implicit pause would be created since it would detect the absence of voice and cease recording. And when the person recording wished to continue pressing a key would signal FS to pass the true audio again therefore reestablishing recording to the same file? Bob

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[POST-1.0.2][RFC][V2 PATCH 0/2] Fine-grained logging: per-session logging
by stkn＠freeswitch.org 25 Dec '08

25 Dec '08

Hi again, This is the reworked patch series to add per-session logging to the FreeSWITCH core. Changes since V1: - Reworked based on second proposed method and feedback by Anthony - Most / all of mod_sofia has been converted to the new system (Needs a review, not sure if all of these changes are OK) Both patches are compile tested (only). Stefan Stefan Knoblich (2): Session log level #1: core support Session log level #2: convert mod_sofia as an example src/include/private/switch_core_pvt.h | 1 + src/include/switch_core.h | 15 +++ src/include/switch_types.h | 5 +- src/mod/applications/mod_commands/mod_commands.c | 37 ++++++ src/mod/applications/mod_dptools/mod_dptools.c | 19 +++ src/mod/endpoints/mod_sofia/mod_sofia.c | 74 ++++++------ src/mod/endpoints/mod_sofia/sofia.c | 96 ++++++++-------- src/mod/endpoints/mod_sofia/sofia_glue.c | 132 +++++++++++----------- src/switch_core_session.c | 15 +++ src/switch_log.c | 24 ++++- 10 files changed, 263 insertions(+), 155 deletions(-)

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custom variables not allowed anymore on a channel?
by seven du 24 Dec '08

24 Dec '08

Hi list- Some custom variable not get send out by event_socket. Like 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(called_trainers=55) we set some variables(including some custom variables) on a channel A-leg before bridge the B-leg. And can get them from the event socket. It works well at least in release 1.0.1. And I believe it works before trunk version 106XX. And at trunk version 10903, we can't get them. Does any one know what happens? Or is their other new ways to do that( e.g. pull the database)? Thanks. 2008-12-23 18:12:51 [DEBUG] mod_recpld.c:297 switch_payload_play_file() done playing payload file:/usr/local/ freeswitch/sounds/custom/eqenglish/8000/welcome.G729 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute ring_ready() 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(ringback=/usr/local/ freeswitch/sounds/custom/8000/american_ringback_tone.raw) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [ringback]=[/usr/local/freeswitch/sounds/ custom/8000/american_ringback_tone.raw] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(continue_on_fail=true) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [continue_on_fail]=[true] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(hangup_after_bridge=true) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [hangup_after_bridge]=[true] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(origination_caller_id_name=IT26335) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [origination_caller_id_name]=[IT26335] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(origination_caller_id_number=IT26335) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [origination_caller_id_number]=[IT26335] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(system_domain=eqenglish) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [system_domain]=[eqenglish] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(called_trainers=55) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [called_trainers]=[55] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(ignore_early_media=true) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [ignore_early_media]=[true] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute set(call_timeout=15) 2008-12-23 18:12:51 [DEBUG] mod_dptools.c:681 set_function() sofia/ outbound/015810303546 SET [call_timeout]=[15] 2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event() sofia/outbound/015810303546 Command Execute bridge({absolute_codec_string='G729,iLBC@30i'}sofia/eqenglish/ trainer39-55) 2008-12-23 18:12:51 [DEBUG] switch_ivr_originate.c:771 switch_ivr_originate() variable string 0 = [absolute_codec_string=G729,iLBC@30i] 2008-12-23 18:12:51 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/eqenglish/trainer39-55 [41eeef28-d0da-11dd-bf27-3fa173bc0bf8] 2008-12-23 18:12:51 [DEBUG] mod_sofia.c:2495 sofia_outgoing_channel() (sofia/eqenglish/trainer39-55) State Change CS_NEW -> CS_INIT December 23, 2008 10:13: Recieved: Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/outbound/015810303546 Unique-ID: 39c85c4e-d0da-11dd-bf27-3fa173bc0bf8 Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: G729 Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: G729 Channel-Write-Codec-Rate: 8000 Caller-Caller-ID-Name: idapted Caller-Caller-ID-Number: 000 Caller-Network-Addr: 61.152.148.84 Caller-Destination-Number: 015810303546 Caller-Unique-ID: 39c85c4e-d0da-11dd-bf27-3fa173bc0bf8 Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/outbound/015810303546 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1230027158183592 Caller-Channel-Created-Time: 1230027158183592 Caller-Channel-Answered-Time: 1230027165689852 Caller-Channel-Progress-Time: 1230027171872337 Caller-Channel-Progress-Media-Time: 1230027158452062 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Other-Leg-Caller-ID-Name: IT26335 Other-Leg-Caller-ID-Number: IT26335 Other-Leg-Network-Addr: 61.152.148.84 Other-Leg-Destination-Number: trainer39-55 Other-Leg-Unique-ID: 41eeef28-d0da-11dd-bf27-3fa173bc0bf8 Other-Leg-Source: src/switch_ivr_originate.c Other-Leg-Context: default Other-Leg-Channel-Name: sofia/eqenglish/trainer39-55 Other-Leg-Screen-Bit: true Other-Leg-Privacy-Hide-Name: false Other-Leg-Privacy-Hide-Number: false Event-Name: CHANNEL_BRIDGE Core-UUID: 6ab3d9b4-d08a-11dd-bf27-3fa173bc0bf8 FreeSWITCH-Hostname: db1.veecue.com FreeSWITCH-IPv4: 192.168.1.14 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2008-12-23%2018%3A13%3A02 Event-Date-GMT: Tue,%2023%20Dec%202008%2010%3A13%3A02%20GMT Event-Date-timestamp: 1230027182869728 Event-Calling-File: switch_ivr_bridge.c Event-Calling-Function: switch_ivr_multi_threaded_bridge Event-Calling-Line-Number: 823

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FS Dev Appreciation
by Michael Collins 24 Dec '08

24 Dec '08

FYI, If anyone would like to show their appreciation for the FS Dev team please email me off list so we can talk about options. I think we can all agree that Tony & Co. have worked very hard to make FreeSWITCH a success for all of us. Let's see if we can give a little love back to the core team! Thanks, Michael

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FreeSwitch, application or module!?
by Dmitry Mordovin 23 Dec '08

23 Dec '08

Hi Friends! I wish to do something like: - Authorize all sip clients in external DB, for example postgresql. - Control each call session duration, just like prepaid calling card. - LCR, if one voip provider can't connect call into PSTN, try next PSTN supplier. Please help me, i can't understand what is docs i might to read to know how to do it. Could you show me some links and terms. explain it? Thank you so much! DM

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Two way tls authentication
by Mohammad Kurmot 23 Dec '08

23 Dec '08

Hi devs; I had configured freeswitch using tls/srtp and it worked fine, by I need to implement two way authentication, I searched wiki and any possible place for configuration, I failed. Does the freeswitch support two-way tls authentication?, if Yes how I can do it. if the answer is No, can I edit the tls module or even write a new module for two way tls. Best Regards Mohammad E. Kurmot

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