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RingBack for PSTN call
by ankitdoshi 29 Jan '16

29 Jan '16

Hello, I want to set .wav as ringback tone for PSTN call, like caller tune. I have implemented it partially for local calls by getting idea from here, https://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones i have set the param like this, $xml .="<action application=\"set\" data=\"ringback=/usr/local/freeswitch/sounds/en/us/callie/astpp-menu.wav\"/>"; before bridge line in dialplan. This works with local calls , but when i dial PSTN call then i hear only ring tone and not my set .wav audio. Any suggestions are welcome. Thanks. ----- Thanks Ankit Doshi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RingBack-for-PSTN-call-tp7596… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Checking which codecs a SIP-Trunk provides.
by Oivvio Polite 29 Jan '16

29 Jan '16

I've registered with SIP-trunk providers for experimentation. How can I figure out which codecs a SIP-trunk (gateway in FS parlance?) provides? regards, Oivvio Polite

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Async Call
by Shan Randhawa 29 Jan '16

29 Jan '16

Respected all, I need some help . I m using this script to make an outbound call and return the status ,How can i make this asynchronous like using bgapi or using execute on answer and run new script but i will still be needing want info about the session in that new script. this_sess = freeswitch.Session("{ignore_early_media=true,setAutoHangup=false,originate_timeout=20,origination_caller_id_number="..callerid.."}sofia/gateway/proxy-301/"..phno)

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180 then 183
by royj＠yandex.ru 29 Jan '16

29 Jan '16

Hi, all In the call flow: SIP UA 1 --> FS --> SIP UA 2 SIP UA 2 answers 180, then 183 with SDP, then sends media. FS forwards 180, 183 to SIP UA 1, but does not send on media at all. But when SIP UA 2 answers 183 with SDP, then sends media, FS forwards 183 and sends on media. Is there some variable to change behavior so FS will send that media or should I file jira? FreeSWITCH Version 1.7.0+git~20160108T231819Z~32d4e56e1e~64bit (git 32d4e56 2016-01-08 23:18:19Z 64bit) ignore_early_media=false bridge_early_media=true Thanks

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Recording conference channel based
by Deepika Yadav 29 Jan '16

29 Jan '16

Hi, I am developing an application based on Freeswitch conference. I need some functonalities like: 1. record session speaker based, currently I record the complete session by setting "auto-record" parameter in conference.conf.xml, I am using a protocol where at a time only two speakers speak at a time and I want to record audio into files having speech from each speakers separately. 2. If I have some timestamps of the conference, can I split recordings based on those timestamps. Currently, I am trying API "uuid_record" to record session in my java application which originates calls and put them into conference and apply "sox" utility to split it into mono audio files but it requires first the recording to made in stereo mode. any pointers in this direction would be appreciable. Regards, Deepika

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UML to document a call flow
by Luis Daniel Lucio Quiroz 29 Jan '16

29 Jan '16

Maybe OT, maybe no. But I have been using UML as a way to let my non-tech users to specify their complex call flow as clear as possible. so far, it has been a sucess I want to share you this little article so you can use it as well. http://inside-out.xyz/technology/how-to-document-a-call-flow-with-uml.html -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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How to build from source v1.6?
by Grant Bagdasarian 29 Jan '16

29 Jan '16

Hello, I've followed the steps for building from source, but I noticed 1.7 is installed, instead of 1.6. https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie How do I force 1.6 to be installed when building from source? I need to enable/disable some unused modules, and I can't find a way to do it when using the "stable branch" steps. These are the steps I following (they're also on the website) Dependencies: apt-get update && apt-get install -y curl curl https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list apt-get update apt-get install -y --force-yes freeswitch-video-deps-most Compiling: # because we're in a branch that will go through many rebases it's # better to set this one, or you'll get CONFLICTS when pulling (update) git config --global pull.rebase true # then let's get the source cd /usr/src/ git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git cd freeswitch.git ./bootstrap.sh -j ./configure make make install

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Seeking collaborators for FreeSWITCH/Lua scripting toolkit
by Chad Phillips 28 Jan '16

28 Jan '16

Looking for folks in the FreeSWITCH community who would be interested in collaborating on an open source scripting toolkit I've written for FreeSWITCH/Lua. The basic idea is to have ready made solutions to common scripting needs, so as not to reinvent the wheel. The original project was written in 2010. A lot of it still works great, and there are some design flaws that I'd like to work out in the next release. Project code is here: https://github.com/thehunmonkgroup/jester Some background and reasoning for the project here: http://xylil.com/2016/01/20/jester-freeswitch-lua-and-my-quest-for-an-aweso… If anyone is interested in helping, please let me know! Chad

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fyi - DTMF with FlowRoute
by Nathan Neulinger 28 Jan '16

28 Jan '16

I know this has been brought up previously in another thread, but wanted to mention I'm seeing it as well. However - I do have an additional data point. This is the SDP being sent: v=0 o=FreeSWITCH 1453729420 1453729421 IN IP4 131.151.x.x s=FreeSWITCH c=IN IP4 131.151.x.x t=0 0 m=audio 19222 RTP/AVP 102 103 9 0 8 104 106 101 a=rtpmap:102 G7221/32000 a=fmtp:102 bitrate=48000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:104 telephone-event/32000 a=fmtp:104 0-16 a=rtpmap:106 telephone-event/16000 a=fmtp:106 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 I realize that anything other than PCMU is pointless with flowroute. Symptom is seen on both 1.4 and 1.6. Confirming what some others on the thread have said - forcing codec to PCMU (in my case via carrier_gateway lcr table), immediately corrects the issue. It seems like a definite bug on flowroute's side, and will be following up with that on the open case. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Freeswitch, webrtc and mod_verto
by Miha 28 Jan '16

28 Jan '16

Hi, is the any info regarding javascript library usage for client part? tnx miha

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