- FreeSWITCH-users - FreeSWITCH Mailing Lists

Better results from mod_vmd
by Matthew Fong 17 Aug '09

17 Aug '09

I tried emailed Eric, seeking advice on this, but his email (the one in the source code) is bouncing email (invalid user), so thought I would ask here instead. If anyone has eric's new email address, I'd be interesting in it. I did some tests with mod_vmd this afternoon, but I'm only finding about 33% of the voice mail beeps and did have 1 false-positive in my test of 7 voice mail machines. I've recorded the audio of the session in .wav files that were both successful and not, as a comparison. I can upload the .wav files if they would be useful. mod_vmd works great for voicemails of Skype Users, and kall8.com, but has issues dealing with mobile phone carriers. sprint - not successful tmobile - not successful verizon - not successful panasonic home answering machine system - not successful kall8 - SUCCESS skype - SUCCESS I'm wondering if you can recommend a simple fix, like changing some of the constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source file, or if better success requires more complex analysis. Do you have any recommendations on how this might be done? Listening to the .wav's its apparent the beeps are not as loud for the mobile phone carriers as they are with skype and kall8. Any guidance would be greatly appreciated. --matt hello hunter http://www.hellohunter.com voice broadcasting & hosted dialer

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Maxmium Number of Concurrent Sessions on EIDE Drive
by Matthew Fong 16 Aug '09

16 Aug '09

I'm interested in doing some testing on the accuracy of mod_vmd (and mod_amd) but wanted to see if anyone could provide some guidelines on the maximum number of concurrent sessions I can record audio files to disk with a typical EIDE drive under 64-bit linux without overloading my system. Also, since I only need to record the incoming audio, would it be suggested I use the api command uuid_record or session:record? Is there a way to only record inbound audio with session:record? Thanks. --matt hello hunter corp. http://www.hellohunter.com hosted dialer & voice broadcasting

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BLF and Openzap
by Terry Moore-Read 16 Aug '09

16 Aug '09

Is it possible to have a sip phone show blf status for a phone which is connected to an openzap port ?

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Is there any freeswitch show version command
by Muhammad Shahzad 16 Aug '09

16 Aug '09

Hi, How can we check the version / svn revision of a running FS instance? I know, this is kind of a stupid question, but i sometimes run into situation where i don't know or don't have access to FS source, nor i can restart it to get its version string. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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ClueCon2009 Torrents
by Peder 16 Aug '09

16 Aug '09

I created torrents for all of the presentations from ClueCon 2009 and have them seeded on 2-3 machines. If you want the torrents, email me off list. Peder

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mod nibblebill question
by Manu 16 Aug '09

16 Aug '09

Hello, If we use heartbeat option on in nibblebill.conf.xml does that mean ODBC database table will be updated every microsecond or any other interval we set? If this is so and there are many users (Lets say 500 users) are connected to FS wouldn't it create locking issues in DB? Regards, Manoj

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Audio only in one direction when calling FS from Skype
by Scott Torr 16 Aug '09

16 Aug '09

Freeswitch-users, I'm very new to Freeswitch and have installed the following in VMware Server 2.0.1 ubuntu-8.04.3-server-i386.iso (udate/upgrade) skype-debian_2.0.0.72-1_i386.deb FreeSWITCH Version 1.0.trunk (14492) mod_skypiax Using the FS wiki to install/setup and making small changes to default XML configs as required. (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via PSTN) The MS-Skype client is on the same local network as FS. I can make calls from the phone to the Skype Client and this works OK. Audio path OK: phone<-->FS<-->Skype But ,if I call from the MS-Skype client to the phone I hear no audio from the MS-Skype client. Audio path: phone<-->FS-->Skype Likewise a call from a Skype 'online number' can hear for example the default 5000 ivr but the DTMF tones from the PSTN phone are not detected by FS when <action application="start_dtmf" /> is used. Any suggestions or pointers in the right direction would be much appreciated. Most likely I have overlook something very obvious to others. Thanks in advance, Scott Torr

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opal build error
by Seven Du 16 Aug '09

16 Aug '09

Hi, According to wiki it still in development status, but should compile right? Any idea about this? thanks. make In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)’ /usr/include/opal/opal/localep.h:267: error: overriding ‘virtual ptlib_virtual_function_changed_or_removed****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)’ mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall&, FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)’ /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h: 276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) mod_opal.cpp: In member function ‘switch_status_t FSConnection::receive_message(switch_core_session_message_t*)’: mod_opal.cpp:1037: error: ‘SWITCH_CHANNEL_SESSION_LOG’ was not declared in this scope make[1]: *** [mod_opal.lo] Error 1 make: *** [all] Error 1

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obtain dialplan content from switch_extension_t
by mark morreny 16 Aug '09

16 Aug '09

Hi, After I do some operation on switch_extension_t, basically adding some actions into it, I would like to display its content in the form of: <extension name="xxx"> <condition> <action>xxxx</action> </condition> </extension> Is that doable? If so, is there any existing functional call that I can use? Thanks, Mark

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Not receiving DTMF
by Nicolas Brenner 16 Aug '09

16 Aug '09

Hi, I'm trying to get dtmf input, but I'm not getting anything. What I discovered though, is that my provider is at fault, since when I switched to another voip provider, everything started to work beautifully. My question is: since my provider is not doing RC2833 dtmf (even though they say they do), is there another way to get dtmf to work? I'm doing everything in a javascript file, so I tried doing: - session.setVariable("dtmf_type","info"); Also: - session.setVariable("dtmf_type","rfc2833"); And: - session.execute("start_dtmf"); But none worked. The voip provider that works would be ideal, but the calls are twice as expensive, hence besides testing, I wouldn't use them for a real case scenario. BTW, anyone know of a good quality VoIP provider with low rates for termination to Santiago, Chile and Chilean cell phones? Thanks! Nicolas

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