November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

LUA error won't start
by Charles Bujold 28 Nov '13

28 Nov '13

Tried the 2 commands, no success. Running version: FreeSWITCH (Version 1.5.7b git cec4b78 2013-11-14 01:42:52Z 64bit) Any other suggestion? Thanks cjb <http://lists.freeswitch.org/pipermail/freeswitch-users/2013-November/101226 .html>

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No password prompt for Voicemail
by Siri MM 28 Nov '13

28 Nov '13

Hi, I am trying to access voicemail in the public context, as my extensions don't authenticate with FS. The problem that I am facing is, when I dial the Voicemail main extension number, I am only prompted for ID, but not password - so I can actually access any extension's Voice mail. My voicemail extension is: <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check auth default ${domain_name}"/> </condition> </extension> My extension is: <include> <user id="3402"> <params> <param name="password" value="g?QaYmAh3a"/> <param name="vm-password" value="1234"/> <param name="vm-enabled" value="true"/> <param name="directory-exten-visible" value="true"/> <param name="dial-string" value="{sip_invite_domain=${domain_name},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="toll_allow" value=""/> <variable name="accountcode" value="3402"/> <variable name="user_context" value="default"/> <variable name="directory-visible" value="true"/> <variable name="limit_max" value="5"/> </variables> </user> </include> Any pointers on what I am missing? Thanks!

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Re: [Freeswitch-users] Apply Connection Charges on LegB
by Shahzad Bhatti 28 Nov '13

28 Nov '13

thanks Avi for the reply! nibblebill is working fine but i want to apply connection charges on leg B when it is answered; i want to apply extra charges as connection charges on legB answer like if call duration is 60 sec rate is 30cents connection charges is 5 cents then call charged as 35 cents 30cent for call duration and 5cents as connection charges i hope that will give idea what i want to say Regards Shahzad Bhatti > > ---------- Forwarded message ---------- > From: Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Thu, 28 Nov 2013 21:05:45 +0500 > Subject: [Freeswitch-users] Apply Connection Charges on LegB > Hi all, i am using mod_xml_curl to bridge call to legB and charge call > using mod_nibblebill and it works fine, but now i want to apply connection > charges to that bridge call when legB is answered but want to know how? > any suggestion will be appreciated > > here is sample of my xml_dialplan > > <?xml version="1.0" encoding="UTF-8" standalone="no"?> > <document type="freeswitch/xml"> > <section name="dialplan" description="Dial Plan For FreeSwitch"> > <context name="default"> > <extension name="Bridge Call to Destination"> > <condition field="destination_number" expression="^1004$"> > <action application="set" > data="process_cdr=b_only"/> > <action application="set" > data="hangup_after_bridge=true"/> > <action application="set" data="nibble_account=1"/> > <action application="set" data="nibble_rate=0"/> > <action application="export" > data="nolocal:nibble_account=1"/> > <action application="export" > data="nolocal:nibble_rate=0.034155"/> > <action application="answer" /> > <action application="bridge" > data="sofia/external/1004%192.168.0.100"/> > <action application="hangup"/> > </condition> > </extension> > <extension name="hangup"> > <condition field="destination_number" > expression="^(hangup)$"> > <action application="respond" data="503"/> > <action application="hangup"/> > </condition> > </extension> > </context> > </section> > </document> > > > Regards > > Shahzad Bhatti > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 28 Nov 2013 16:21:06 +0000 > Subject: Re: [Freeswitch-users] Apply Connection Charges on LegB > Since nibblebill only charges for answered calls, that looks like it > should work. > Is it not working as expected? > -Avi > >

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Apply Connection Charges on LegB
by Shahzad Bhatti 28 Nov '13

28 Nov '13

Hi all, i am using mod_xml_curl to bridge call to legB and charge call using mod_nibblebill and it works fine, but now i want to apply connection charges to that bridge call when legB is answered but want to know how? any suggestion will be appreciated here is sample of my xml_dialplan <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="Dial Plan For FreeSwitch"> <context name="default"> <extension name="Bridge Call to Destination"> <condition field="destination_number" expression="^1004$"> <action application="set" data="process_cdr=b_only"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="nibble_account=1"/> <action application="set" data="nibble_rate=0"/> <action application="export" data="nolocal:nibble_account=1"/> <action application="export" data="nolocal:nibble_rate=0.034155"/> <action application="answer" /> <action application="bridge" data="sofia/external/1004%192.168.0.100"/> <action application="hangup"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="respond" data="503"/> <action application="hangup"/> </condition> </extension> </context> </section> </document> Regards Shahzad Bhatti

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PDD Math
by Andre 28 Nov '13

28 Nov '13

How do I calculate PDD in the freeswitch CDR's? Per http://wiki.freeswitch.org/wiki/PDD In Seconds: * progress_mediasec * progresssec In milliseconds (1000 = 1 second): * progress_mediamsec if there was media * progressmsec if there was no media In MicroSeconds (1000 = 1 millisecond) * progress_mediausec * progressusec but most of my calls that are not answered are 0 on both A and B legs.

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rtp-timer-name param
by Ashwin Jain 28 Nov '13

28 Nov '13

What is "rtp-timer-name"? -- Thanks and Regards, Ashwin Jain Director (Engineering) MetroGuild

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disastery plan
by Nikita Tarasov 28 Nov '13

28 Nov '13

Hi guys. We have loadbalancer(opensips) and a huge amount of incoming calls going to ONE sip server, which is unscalable and priopretary. I need to create some kind of disastery plan for this server service is down or on high load. The main idea of this plan is, when disastery happends, to balance all incoming calls to freeswitch and keep them with music, while main server is down. On recovery calls should be forwarded to main server with the same destination number as they was. I have tryed to make it with mod_callcenter. It was done by placing calls in queue and when main server is on and re-registered on freeswitch, calls was forwarded. But without destination number. And I can't copy destination number to forwarded calls. Can you help me?

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Sound files for mod_say_he
by Zev Spitz 28 Nov '13

28 Nov '13

According to the link at http://wiki.freeswitch.org/wiki/Mod_say_he, the sound files for mod_say_he will be available soon. The page was last updated 4 April 2011. Has anything happened since? Thanks in advance, Zev Spitz

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LOL - What language should I use?
by James Mortensen 27 Nov '13

27 Nov '13

All joking aside, it's clear that the author in the FreeSWITCH docs prefers Lua. It sounds like, hands down, Lua is the answer based on information found here: http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F But why isn't Java in the list? Is Lua the fastest scripting language when only compared to Perl, Python, and JavaScript? What types of results have folks compiled from tests with mod_java? I ask because this seems to be a bit of a hole in the docs. Java is by far one of the most widely used languages (although I personally prefer something other than Java) so it seems that this should be covered too. Hope this helps and thank you! James

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pablio.h - MAX_IO_CHANNELS hardcoded to 2
by Erik M. Devane - Comms Guy 27 Nov '13

27 Nov '13

Dear all, I've been fighting an issue with PortAudio on CentOS for a few days now: I have two Delta 1010 cards which I'm using with mod_portaudio streams. On Windows, the cards appear as eight two-channel devices, and FreeSWITCH can use them. On CentOS, the cards appear as two eight-channel devices (actually 10 out/12 in) and I'm unable to configure more than two channels per card. Looking at the code, it looks like pablio.h's MAX_IO_CHANNELS is hardcoded to 2, and this is the limit of mod_portaudio. As it's a ring-buffer, I can see why it might need to be a power of 2, but does anyone know why it is set to 2? How dangerous would it be to increase this value? Any thoughts? Regards, Erik

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