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mod_unimrcp dont send invite while using a Mrcp v2 profile
by Leonardo P. Bidinoto 21 May '13

21 May '13

Hi All, Im new at using the mod_unimrcp. I'm having some troubles when configuring to using several profiles at same time. I have configured 3 profiles(2 with Nuance and 1 Verbio, all using MRCPv2). But when i try to use a TTS or ASR from any of those profile, FS channel get stuck and i get a warning message telling that the time for the MRCP server to respond have expired. I have used wireshark to capture the packages, and when i call speak or detect_speech function of FS, i dont see any SIP INVITE going from FS to the MRCP Server. Did i miss something at configuring profiles? Did anyone get something similar to this situation? I'm using the examples from FS mod_unimrcp page. Any tip will help a lot. Thanks -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br

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GXP1405 G729 audio problem
by Juan Antonio Ibañez Santorum 21 May '13

21 May '13

Hello, Anyone had problems using G279 codec with Grandstream GXP1405. GXP1405 is connected to FS which is connected to an Asterisk box. FS has no codec installed so it is in passthrough mode. The audio received in the phone is bad. Sometimes volume goes down and it has low quality. Working connected directly to Asterisk using G729 box goes well. Working through FS using G711 goes well. If I analyze both flows, from Asterisk and from FS, inside FS box using G729, all audio is OK. So I don't know why phone plays it bad... Any idea? Regards

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Re: [Freeswitch-users] custom sip header not working in dialplan
by windy 21 May '13

21 May '13

Hi, Freeswitch Team There is problem for to modify the outbound sip header. As the freeswitch wiki says, it can modify/add custom sip header via below method: <action application="set" data="sip_h_X-Charge=1"/> <action application="set" data="sip_rh_X-Reason=Destination Number Not in Footprint"/> <action application="set" data="sip_bye_h_X-Accounting=Some Accounting Data"/> Now, I add the red lines in dialplan/default.xml. But only the custom header prefixed with "sip_h_" can be shown in the SIP INVITE message. Other custom headers prefixed with "sip_bye_h" and "sip_ph"... not found in the related messages(BYE, 183). Could you kindly give comments about this case? <context name="from-pstn-to-sip"> <extension name="to-sip">  <condition field="destination_number" expression="^(.{1,})$" break="never"> <action application="set" data="destnumber=$1"/> <anti-action application="set" data="destnumber=unknown"/> </condition>  <condition field="destination_number" expression="^(.*)$"> <action application="set" data="hangup_after_bridge=yes" /> <action application="set" data="tone_detect_hits=1" /> <action application="export" data="fax_enable_t38_request=true" /> <action application="export" data="fax_enable_t38=true" /> <action application="tone_detect" data="faxdisable 1100 r +5000 disable_ec 1"/> <action application="export" data="execute_on_answer=tone_detect fax_disable_ec 2100 r +5000 t38_gateway 'self nocng'" /> <action application="set" data="sip_contact_user_replacement=${destnumber}"/> <action application="set" data="hangup_after_bridge=yes"/>  <action application="export" data="sip_info_when_no_2833=false" /> <action application="set" data="sip_h_X-Charge=1"/> <action application="set" data="sip_rh_X-Reason=Destination Number Not in Footprint"/> <action application="set" data="sip_bye_h_X-Accounting=Some Accounting Data"/>    <action application="bridge" data="sofia/external/sip:${destination_number}@172.20.2.17:5062"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> Thanks Windy

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Problem with enable-post-var
by Anatolii 21 May '13

21 May '13

Hello all. In mod_xml_curl in binding configuration I try make dynamic "gateway-url". For example, <param name="gateway-url" value="http://hostname/configuration/<b>$${my_var}*" bindings="configuration"/> Also try to add 'enable-post-var' -- didn't work too. Please, help anyone. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-enable-post-var-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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UUID - how unique is it?
by Telecube - John 21 May '13

21 May '13

How unique is uuid in the cdr records? Coming from asterisk where uid is about as unique as a black bowling ball.... Can I count on uuid to be totally unique in my billing records? -- Regards, John Matich Telecube Pty Ltd www.telecube.com.au <http://www.telecube.com.au> Ph: 13CUBE (132823)

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FreeSWITCH RTMFP support
by Adam Raszynski 21 May '13

21 May '13

Hi Are there any plans to add support form Adobe RTMFP protocol? I know that there is mod_rtmp, but it supports only RTMP which is TCP-based. RTMFP is UDP based and designed specially for P2P communications and should give much smaller delays Kind Regards

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FreeSWITCH Weekly News and Notes
by Michael Collins 20 May '13

20 May '13

Hello! The FreeSWITCH team is busy on several fronts but the most exciting news at the moment is that Packt Publishing has told us the new book<http://www.packtpub.com/freeswitch-1-2/book>is officially off to production! If all goes well it should be available in the next few weeks. (It looks like they changed the cover art - once we know for sure what's on the cover we'll give it a nickname like we did with the "bridge" book.) Last week we had an impromptu discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_15>on various FreeSWITCH topics. We have rescheduled the VoIPMonitor.org conference call presentation for May 29th. Also, we have tentatively scheduled the HOMER presentation for June 5. That means we will have another open discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22>for this Wednesday's conference call. As a tip, if you would like to browse around and see what has been committed in FreeSWITCH lately you can use the Fisheye site<http://fisheye.freeswitch.org/>. Here you can browse the source code and look at recent commits to see what has changed. If you prefer to look at the commits without the source code browser then try out the HTML front-end <http://fisheye.freeswitch.org/>for our FreeSWITCH git repository. We also have repositories for contributions <http://git.freeswitch.org/git/freeswitch-contrib/> and sample configuration <http://git.freeswitch.org/git/freeswitch-sample-configs/>sets. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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directory sql tables
by adahary 20 May '13

20 May '13

I planning to switch from the directory xml files to sql tables using mod_xml_curl.I've already defined ODBC/Mysql and got all core sql tables installed.I've search for some sql tables for directory with all params/variables but couldn't find any.With Astwerisk RT there is a clear WiKi which layout all sql tables with user/system fields.Are there any ready to use directory sql tables? OR should I make my own tablesRegardsAssaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/directory-sql-tables-tp759087… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Thoughts on security/code injection/etc. in FS when allowing user supplied data
by Nathan Neulinger 20 May '13

20 May '13

I've noticed several places in FS code and examples where it isn't safe at all to take user supplied data. An easy example is the use of mailer_app: #ifdef WIN32 switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < \"%s\"\"", runtime.mailer_app, from, runtime.mailer_app_args, to, filename); #else switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s %s %s", filename, runtime.mailer_app, from, runtime.mailer_app_args, to); #endif another is ANY use of passing channel vars or data to a system or bgsystem command. This isn't an issue normally, but if you want to give limited ability for users to control their own dial rules, then you wind up having to be very careful with processing the data to make sure it's safe. That's always a good idea, but it still seems like a bad idea to take that data and then directly use it in a completely unsafe context like a parsed command line. For the voicemail notify case, seems like an easy answer would be something like a "vm-notify-hook", which at that point, could call out to lua or perl to do the actual sending in a safe manner, passing the recipient/sender/etc. as data instead of on cmd line. For the 'passing channel vars...' case, I think it would be good to have a 'system_json' and 'bgsystem_json' set of routines that would pass channel data to the script on stdin in json format. Regardless of implementation of either of those, I think it would be worthwhile to have a shell_escape() routine in the core utilities to allow the current syntax to be used more safely. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Recommendation for GUI to PBX
by Cesar Bermudez 19 May '13

19 May '13

Hi all. What its the best GUI for use FS in pbx? and why? bluebox? fusion? any other? Regards.

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