May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Kernel frequency
by peely 04 May '12

04 May '12

Hi, I run my production stuff on Ubuntu Server and have been using the pre-emptive kernel which runs at 1000hz, this was because previous build of FreeSWITCH seemed to need to run at this frequency. In testing the latest Ubuntu LTS release, it seems they're not producing recent kernels for anything other than their generic Kernel, which runs at 250hz I believe. Is it still a requirement / recommendation that FreeSWITCH runs on 1000hz boxes? I know the preferred distro is Centos, but would appreciate some guidance on what may occur if I were to move to 250hz kernels on production boxes, obviously this would still be a "softly softly" move. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Kernel-frequency-tp7528206.ht… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH-biz Mailing List
by Michael Collins 04 May '12

04 May '12

Hello all, I'd like to encourage everyone to read this post<http://freeswitch.org/node/392>on the main FreeSWITCH.org site. In short: we are finding that there are more and more business opportunities for members of the FreeSWITCH community. Those with marketable FreeSWITCH skills are increasingly in demand. One way to keep everyone informed about these opportunities is to have a place where people can post and discuss these opportunities. The ideal place for this is the FreeSWITCH-biz mailing list<http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz>. We encourage everyone to subscribe to this list so that we can keep business discussions separate from user and developer questions. -Michael

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FreeSWITCH Conference Call: Vestec ASR
by Michael Collins 04 May '12

04 May '12

Hello all! Just a reminder that the FreeSWITCH community conference call will be starting in just over an hour. The agenda page is here<http://wiki.freeswitch.org/wiki/FS_weekly_2012_05_02>. Come join us for an interesting talk with the Vestec team about automatic speech recognition (ASR) in a FreeSWITCH environment. -Michael

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Re: [Freeswitch-users] odbc with FreeTDM
by Peter Olsson 04 May '12

04 May '12

AFAIK there is not ODBC in FreeTDM - what are you trying to do? /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anita Hall Skickat: den 4 maj 2012 10:41 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] odbc with FreeTDM Hi Could anyone share a sample config of odbc with FreeTDM ? Thanks. regards, Anita !DSPAM:4fa3951932765625512396!

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odbc with FreeTDM
by Anita Hall 04 May '12

04 May '12

Hi Could anyone share a sample config of odbc with FreeTDM ? Thanks. regards, Anita

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Call forwarding to mobile showing our trunk CLI instead of caller CLI
by Neil Stirton 04 May '12

04 May '12

Hi, Apologies for the noob question here. Setup Freeswitch on CentOS and using FusionPBX for 'easy' management. I've defined a hunt group (ext 7001) which simultaneously rings 3 internal SIP extensions + my mobile. All the internal extensions see the caller ID no problem but because I am also sending the call back out through our ITSP's trunk, the call always shows as our office calling my mobile phone rather than the actual caller's ID. I have the hunt group calling: Sip uri sofia/gateway/<gateway>/<my-mobile> I'm sure it's a case of a variable setting something but if someone could point me in the right direction that would be very much appreciated. Many thanks Neil

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Early media issue
by Cucu Florian 04 May '12

04 May '12

Hello, I have the following problem regarding early media. When I call from Xlite(extension 1000) to one of my SIP endpoint(extension 1001) I hear a fake long ring instead of media. Both endpoints accept G711ulaw and alaw codec. I think that 1001 send an 183 SIP response that is misinterpret by FS or Xlite. Is any solution to hear in xlite the correct media that 1001 send it? Thanks in advance. My Freeswitch logs are: 2012-05-03 23:40:25.027591 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1>1001 in context default 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *1 ee_extension::dx XML features 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *2 r_session::/usr/local/freeswitch/recordings/1000.2012-05-03-23-40-25.wav 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *3 ee_extension::cf XML features 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *4 ee_extension::att_xfer XML features 2012-05-03 23:40:25.087371 [NOTICE] switch_channel.c:926 New Channel sofia/nal/sip:1001@192.168.1.69:5060 [35ddd1a6-9560-11e1-8f20-dd25b25b02e6] 2012-05-03 23:40:25.187367 [NOTICE] sofia.c:5633 Pre-Answer sofia/internal/ 001@192.168.1.69:5060! *2012-05-03 23:40:25.210095 [INFO] switch_ivr.c:725 Sending early media* *2012-05-03 23:40:25.210095 [NOTICE] mod_sofia.c:2578 Pre-Answer sofia/ inter000(a)192.168.1.10!* 2012-05-03 23:40:56.610515 [NOTICE] sofia.c:6293 Hangup sofia/internal/ sip:1001@192.168.1.69:5060 [CS_HIBERNATE] [NORMAL_TEMPORARY_FAILURE] 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1398 Session 22 (sofia/internal/sip:1001@192.168.1.69:5060) Ended 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1001@192.168.1.69:5060 [CS_DESTROY] 2012-05-03 23:40:56.610515 [NOTICE] mod_dptools.c:1135 Channel [sofia/internal/1000(a)192.168.1.10] has been answered 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:926 New Channel loopback/app=voicemail:default 192.168.1.10 1001-a [494485e6-9560-11e1-8f2c-dd25b25b02e6] 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:924 Rename Channel loopback/app=voicemail:default 192.168.1.10 1001-a->loopback/voicemail-a [494485e6-9560-11e1-8f2c-dd25b25b02e6] 2012-05-03 23:40:57.647627 [NOTICE] switch_channel.c:926 New Channel loopback/voicemail-b [49452e10-9560-11e1-8f30-dd25b25b02e6] 2012-05-03 23:40:57.647627 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2012-05-03 23:40:57.647627 [NOTICE] mod_dptools.c:1161 Pre-Answer loopback/voicemail-b! *2012-05-03 23:40:57.647627 [WARNING] switch_ivr_bridge.c:246 Cannot bypass media while bridged to a loopback address.* *2012-05-03 23:40:57.647627 [ERR] switch_core_io.c:131 sofia/internal/ 1000(a)192.168.1.10 has no read codec.* 2012-05-03 23:40:57.647627 [NOTICE] switch_core_io.c:132 Hangup sofia/internal/1000(a)192.168.1.10 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2012-05-03 23:40:57.667365 [NOTICE] switch_ivr_bridge.c:667 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2012-05-03 23:40:57.667365 [NOTICE] mod_loopback.c:438 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1398 Session 21 (sofia/internal/1000(a)192.168.1.10) Ended 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/1000(a)192.168.1.10 [CS_DESTROY] 2012-05-03 23:40:57.719451 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_play_greeting]: '1001' did not match any patterns 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 23 (loopback/voicemail-a) Ended 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close Channel loopback/voicemail-a [CS_DESTROY] 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 24 (loopback/voicemail-b) Ended 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close Channel loopback/voicemail-b [CS_DESTROY]

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can't build mod_gsmopen ?!
by Tamer Higazi 03 May '12

03 May '12

Hi people! I am on gentoo, and fetched through it's overlay with git the snapshot and tried to build FS with mod_gsmopen. During built process it stoped with this error: making all mod_gsmopen copying header files quiet_libtool: link: warning: `/usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../lib64/libodbc.la' seems to be moved Compiling gsmopen_protocol.cpp... In file included from gsmopen_protocol.cpp:1:0: gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h: Datei oder Verzeichnis nicht gefunden Kompilierung beendet. make[5]: *** [gsmopen_protocol.o] Fehler 1 make[4]: *** [all] Fehler 1 make[3]: *** [mod_gsmopen-all] Fehler 1 make[3]: *** Warte auf noch nicht beendete Prozesse... Any advises how to solve that?! Would be really cool to connect the cell through bluetooth and route it to my A100 board. Tamer

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Javascript - new V8 Engine?
by Avi Marcus 03 May '12

03 May '12

I know that lua has better performance and "embeddability" than any other language for IVRs and other scripts. If mod_javascript would be upgraded to use V8, would that make using javascript on par with Lua? If so.. what size bounty would push it along? Thanks! -Avi

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spandsp consultant T.30
by Anita Hall 03 May '12

03 May '12

Hi One of our clients is getting around 70% results using mod_spandsp on FreeSWITCH. The protocol is T.30 and the client is India. They are looking for someone who could help them to increase the results to more than 95%. If you or someone you know could help, please let me know. Thanks! regards, Anita

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