- FreeSWITCH-users - FreeSWITCH Mailing Lists

Having problems (no sound) when trying to LogIn from Home to Freeswitch
by Joao Leme 18 Dec '10

18 Dec '10

Hi there! First time here. I'm having problems (no sound) when trying to LogIn from Home to Freeswitch. In the office it all works fine, even when I use the external IP as Domain on X-Lite. I'm trying to go for internal profile on port 5060 as I want to be just like if I was there in the office. It actually works, I can make calls, receive etc but NO SOUND. What I also did was to change the domain on var.xml to the external ip. I also forwarded the ports to the computer where Freeswitch according to Firewall info on docs. Thanks a lot, John

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FS G729 Release: 147
by Brian West 17 Dec '10

17 Dec '10

New release of FSG729 that fixes a memory leak has been released: http://files.freeswitch.org/fsg729-147-installer Added API options are: ${g729_count()} returns the number of permitted licensed channels. ${g729_available()} returns true or false if the codec is loaded and license server is functional. ${g729_used()} returns the encoder:decoder counts. This paired with ${regex()} can be powerful. g729_status has been replaced with g729_info. Please provide feedback if you run into problems. Thanks, Brian

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FS ends call on DTMF **
by Cliff Wells 16 Dec '10

16 Dec '10

Hi, I've got a Lua application that takes the caller id and generates particular sequences of DTMF tones (for a testing system) based on the CID. If I call into the system (using my cellphone), it correctly plays the sequence. However, if I press ** on the keypad it stops the generated DTMF tones and then apparently hangs up. This wouldn't be a huge issue (the app isn't supposed to receive any DTMF anyway) except that one of the systems calling in apparently has enough echo on the remote end that the DTMF I'm generating is being echoed back to FS and the first two DTMF tones I generate are, of course, "**", so the end result is the system calls into FS, gets two tones and the call is terminated. As an aside, it appears that ** is unique in this way. If I dial "99" or "77", the DTMF still pauses, but then resumes (or maybe restarts... it's difficult to tell). I was using an older version of FS, so I did "make current" a couple of hours ago, but it didn't help. My one and only dialplan is this: <extension name="responder"> <condition field="destination_number" expression="^(.*)$"> <action application="lua" data="responder.lua"/> </condition> </extension> And responder.lua does this: session:answer () session:sleep (3000) session:execute ("playback", genstream (val)) -- genstream uses the CID to produce a tone_stream session:hangup() (clearly there's a bit more to the Lua app, but nothing relevant to this issue). I see this in the console when this happens: 2010-12-17 01:04:34.603404 [NOTICE] mod_dptools.c:920 Channel [sofia/internal/3232269108(a)99.99.99.99] has been answered EXECUTE sofia/internal/3232269108(a)99.99.99.99 lua(responder.lua) 2010-12-17 01:04:34.673314 [DEBUG] sofia.c:4606 Channel sofia/internal/5551231234(a)99.99.99.99 entering state [ready][200] 2010-12-17 01:04:34.751303 [DEBUG] switch_rtp.c:2657 Correct ip/port confirmed. EXECUTE sofia/internal/5551231234(a)99.99.99.99 playback(tone_stream://*(200,200);*(200,200); ... 2010-12-17 01:04:37.915169 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms 2010-12-17 01:04:38.295137 [DEBUG] switch_rtp.c:3033 RTP RECV DTMF *:1440 2010-12-17 01:04:38.295137 [DEBUG] mod_dptools.c:1634 Digit * 2010-12-17 01:04:38.295137 [DEBUG] switch_ivr_play_say.c:1573 done playing file 2010-12-17 01:04:38.295137 [DEBUG] switch_cpp.cpp:602 CoreSession::hangup 2010-12-17 01:04:38.295137 [DEBUG] switch_channel.c:2455 (sofia/internal/5551231234(a)99.99.99.99) Callstate Change ACTIVE -> HANGUP 2010-12-17 01:04:38.295137 [NOTICE] switch_cpp.cpp:604 Hangup sofia/internal/5551231234(a)99.99.99.99 [CS_EXECUTE] [NORMAL_CLEARING] 2010-12-17 01:04:38.295137 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/5551231234(a)99.99.99.99 [KILL] 2010-12-17 01:04:38.295137 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/5551231234(a)99.99.99.99 [BREAK] 2010-12-17 01:04:38.295137 [DEBUG] switch_cpp.cpp:972 sofia/internal/5551231234(a)99.99.99.99 destroy/unlink session from object 2010-12-17 01:04:38.295137 [DEBUG] switch_core_session.c:1998 sofia/internal/5551231234(a)99.99.99.99 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) At this point I'd be content with a simple workaround (mute the inbound audio, ignore DTMF, etc), since I do not need any feedback from the other end. Cliff Wells <cliff(a)develix.com>

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Calling Card System Setup - Suggestions for New Setup?
by Avi Marcus 16 Dec '10

16 Dec '10

Hi, I'm setting up a calling card system with FreeSWITCH. Figured I'd ask before setting it all up- Does anyone have any suggestions, or scripts, that they use for auto-login or for logging in users to check their balance? Any specific pitfalls to worry about? I figured I could bypass media once the user places their call. Is that a bad idea? And.. any suggestions for payment processors other than paypal - some people signing up won't have internet, their CC will have to be manually keyed in. I've heard one place suggest authorize.net Thanks! -Avi

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FreeSwitch / Solaris 10 / Sun Studio 12
by Marty Lee 16 Dec '10

16 Dec '10

Current GIT repository doesn't compile on Solaris 10 with Sun Studio due to unresolved link errors on the 'make_mask32' function. I've isolated the function and added it to the libfreeswitch library - this cures the link problems. Patches attached if anyone is interested. m ----- Marty Lee e: marty(a)maui-systems.co.uk Technical Director v: +44 845 869 2661 Maui Systems Ltd f: +44 871 433 8922 Scotland, UK w: http://www.maui-systems.co.uk

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SIP MESSAGE requests
by John Rose 16 Dec '10

16 Dec '10

Hello, Does anyone know if it is possible to construct, originate and receive SIP MESSAGE requests in FreeSwitch or libfreeswitch? This would be in a sessionless dialog as of RFC 3428 Page mode. Thanks, John

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SqlCommand could not be found: mod_managed compile error in CentOS
by srinivasula reddy 16 Dec '10

16 Dec '10

Hi All, When i am trying to build mod_managed in Freesiwtch Latest version, i am getting this error, this error i got in my own class created for updating SqlServer Database, any idea? -- Srinivasula Reddy K

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mod_conference wiki error
by Madovsky 16 Dec '10

16 Dec '10

<action application="conference" data="confname@profilename+flags{mute|deaf|waste|moderator}+[conference pin number]"> should be<action application="conference" data=confname@profilename+[conference pin number]+flags{mute|deaf|waste|moderator}>

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Re: [Freeswitch-users] mod_conference wiki error
by Madovsky 16 Dec '10

16 Dec '10

Done... Don't wait for authorisation. Anything that's found to be wrong can always be undone or corrected later. Steve on iPhone ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Thursday, December 16, 2010 9:43 AM Subject: Re: [Freeswitch-users] mod_conference wiki error I didn't want without your authorization... ----- Original Message ----- From: Brian West To: FreeSWITCH Users Help Sent: Thursday, December 16, 2010 9:23 AM Subject: Re: [Freeswitch-users] mod_conference wiki error The beauty of the wiki is you can update it yourself... have you done so already? /b On Dec 16, 2010, at 12:31 AM, Madovsky wrote: <action application="conference" data="confname@profilename+flags{mute|deaf|waste|moderator}+[conference pin number]"> should be<action application="conference" data=confname@profilename+[conference pin number]+flags{mute|deaf|waste|moderator}> _ ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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FreeSwitch / Solaris 10 Zone / Sun Studio
by Marty Lee 16 Dec '10

16 Dec '10

ok, so now I have FreeSwitch up and running in a Solaris 10 Zone, compiled with Sun Studio 12 and two SIP phones attached and passing calls - result! For the developers, the libs/esl/Makefile is not autogenerated and has a load of 'g++' specifics in it; once I adjusted those to the Sun Studio settings, the 'hash' module built fine. Last one; know this isn't your issue, but the 'flite' module uses the csw_sts.h file and defines a number of 'const' variables in structures. The code to do a '#define const' to disable those seems to be being applied to the system includes too, resulting in a number of function conflicts - the ones that preceed the #define and any that are nested that follow. Instead of trying to 'bodge' the code to disable 'const', it was much easier to just remove the 'const' from the csw_sts.h file itself. Again, more ramblings in case any other 'Solaris' people ever end up here. BTW, this has just been tested within a Solaris Zone - something that I can't get to work on Asterisk, which was my main reason for looking for an alternative. If I can get IAX and SIP working reliably, then this will become the next prod' solution. Right - onwards and inwards to understand the xml file etc. :-) m ----- Marty Lee e: marty(a)maui-systems.co.uk Technical Director v: +44 845 869 2661 Maui Systems Ltd f: +44 871 433 8922 Scotland, UK w: http://www.maui-systems.co.uk

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