- FreeSWITCH-users - FreeSWITCH Mailing Lists

important message
by Vlasis Hatzistavrou 22 Oct '15

22 Oct '15

Hello! New message, please read <http://bargaincarhire.com/off.php?onbj> Vlasis Hatzistavrou

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important message
by Kristin Kahlden 22 Oct '15

22 Oct '15

Hello! New message, please read <http://www.resalahsms.com/laughing.php?hfe83> Kristin Kahlden

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Unable to compile source missing libvpx-dev on CENTOS
by Naveen Tamanam 22 Oct '15

22 Oct '15

Hi, I am unable to compile freeswitch from source. getting the following error, making all mod_vpx make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/codecs/mod_vpx' Makefile:883: *** You must install your distros libvpx-dev to build mod_vpx. Stop. make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/codecs/mod_vpx' make[3]: *** [mod_vpx-all] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [all] Error 2 then after I have installed the libvpx-devel.x86_64 (yum install libvpx-devel) still I am getting same error when I during compilation. -- Thanks & Regards, Naveen Tamanam

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Re: [Freeswitch-users] NDLB for SIP INFO
by Chris B. Ware 22 Oct '15

22 Oct '15

Hi Michale, - I use SIP INFO since I need to bypass media (no RTP stream for local calls).- I reported it to both vendors. Gigaset say they are compliant with many PBX vendors, so they don't need to change it. Yealink should answer soon. Anyway Gigaset IP Cordless are good and cheap, so many customers would like to use them, that's why I'm asking the patch. Thank you. >I don't think it exists. It might be possible to add one and we would >probably review that patch, but why wouldn't you just use rfc2833 instead >and/or report this error to the vendors? > >>On Thursday, October 22, 2015, Chris B. Ware <chrisbware at yahoo.it> wrote: >> Hi guys, >> >> I've discovered that Gigatset and Yealink IP cordless send SIP INFO in a >> wrong way, according to SDP RFC. >> If I press 4, I get on SDP >> >> Signal=4\r\n >> Duration=100 >> >> While should be >> >> Signal=4\r\n >> Duration=100\r\n >> >> >> Freeswitch (1.4.23) answers with 400 Bad Request. I wonder if there's >> something like a NDLB params to accept this format, based on SDP Content >> Length. >> >> >> Thank you >> >>

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NDLB for SIP INFO
by Chris B. Ware 22 Oct '15

22 Oct '15

Hi guys, I've discovered that Gigatset and Yealink IP cordless send SIP INFO in a wrong way, according to SDP RFC.If I press 4, I get on SDP Signal=4\r\n Duration=100 While should be Signal=4\r\n Duration=100\r\n Freeswitch (1.4.23) answers with 400 Bad Request. I wonder if there's something like a NDLB params to accept this format, based on SDP Content Length. Thank you

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FYI - fcc robocaller data feed
by Nathan Neulinger 22 Oct '15

22 Oct '15

http://consumerist.com/2015/10/21/fcc-now-sharing-weekly-robocall-complaint… https://consumercomplaints.fcc.gov/hc/en-us/articles/205239443-Data-on-Unwa… Unfortunately, doesn't look to be a nice fixed URL at the moment: https://consumercomplaints.fcc.gov/hc/theme_assets/513073/200051444/Telemar… -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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rtp_use_dtls vs. webrtc_enable_dtls
by Johannes Singler 22 Oct '15

22 Oct '15

There are two channel variables whose names are very similar: rtp_use_dtls (only regarded when "true") and webrtc_enable_dtls (only regarded when "false") They are not mentioned in the Confluence wiki, both are used switch_core_media.c. So what do they mean exactly, and when should either of them be used (or could they be merged maybe)? Johannes

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Break unaswered bridge
by Alexey Melnichuk 21 Oct '15

21 Oct '15

This is simple dialplan that I try to implement <extension> <condition field="destination_number" expression="^\d+$" break="on-true"> <action application="bind_digit_action" data="conf,*1,exec:execute_extension,xfer"/> <action application="digit_action_set_realm" data="conf"/> <action application="conference" data="100@default"/> </condition> <condition field="destination_number" expression="^xfer$" break="on-true"> <action application="bind_digit_action" data="xfer,*1,?????????"/> <action application="digit_action_set_realm" data="xfer"/> <action application="digit_action_set_realm" data="xfer"/> <action application="read" data="2 11 'tone_stream://%(10000,0,350,440)' target_num 30000 #"/> <action application="bridge" data="user/${target_num}@${domain_name}"/> </condition> </extension> I have two problem with this dialplan 1. in secon `bind_digit_action` I can not use `exec` only `api`. e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/> does not work. Application executes only when I return from first `exec:execute_extension` 2. I can not cancel bridge before answer I can not use transfer because I do not whant leave conference (e.g. user has `end_conference=true` flag). I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}" -- kill_legs.lua local uuid = argv[1] local sql = ("select uuid from channels where call_uuid='%s' and uuid<>'%s'") :format(uuid, uuid) dbh:query(sql, function(row) local res = api:executeString("uuid_kill " .. row.uuid) end) But this is does not work because it kills only active channels, but for dial-string like `loopback/111/my.domain,[leg_delay_start=20]loopback/222/my.domain` second channels creates only after 20 sec. And because I may have more then one channels I can not use `originate_uuid` In documentation I found `break` application but as I say I can not use it. `uuid_break` api break only play audio.

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record_session - dtmf and silence
by Leonardo Lima Ribeiro 21 Oct '15

21 Oct '15

I have a lua script creates a session and do a lot of stuff… And I record this session using the record_session api command… The problem is, sometimes I have session:execute(‘send_dtmf’,’1234@250’)… And in the log, I can see that too, but not at the session wav file. Is it possible to have this dtmf’s at the wav file? The second problem is, sometimes I have a big silence, like 10 seconds of no sound, and at the session wav file I don’t have this silence interval, but it’s important to me… Could I have this silence at the session wav too? Thank you, Leonardo Ribeiro

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In B2BUA mode, how can I delay answering first call
by Dan Edwards 21 Oct '15

21 Oct '15

I am using FreeSWITCH as a B2BUA to handle WebRTC calls and link them to a PBX that does not have WebRTC support. When the PBX calls a WebRTC user, FreeSWITCH answers the initial call, then initiates the call to the WebRTC user. This causes the first 2-4 seconds of the audio to be lost. Is there a way to have the first call wait until the second call is answered? Thank you, Dan

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