January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

disable user features in conference
by Joli Martinez 23 Jan '14

23 Jan '14

Hello, When a user logs into the conference he is able to press * to mute, # to kick himself out, and other options. I would like to disable them all and let me manage the state of the user. Where would I disable those options? Thanks,

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call rejected due to min_idle_cpu overload counter
by Alain BIEUZENT 23 Jan '14

23 Jan '14

Hi All, is there a way to know the number of calls that have been rejected by exceeding the value of min_idle_cpu ? Regards Alain.

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trans-coding is not working
by Shahzad Bhatti 23 Jan '14

23 Jan '14

Hi Everybody, i want to call legB in a scnerio that i have "inbound-codec-prefs" value="GSM,PCMA,PCMU,G729" "outbound-codec-prefs" value="PCMA,PCMU,G729" so i change sip profile as <param name="inbound-zrtp-passthru" value="false"/> <param name="inbound-late-negotiation" value="true"/> then after *reload mod_sofia* i originate call to bLeg where on inbound i only offer GSM Codec and i think that GSM codec should trans code using PCMA but i got response [415] here is my cli log http://pastebin.freeswitch.org/21881 tell me how i can trans code and correct me if i am wrong; Regards Shahzad Bhatti

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no video when bridge call to PBX ACD
by Jimmy Chang 23 Jan '14

23 Jan '14

Hi folks, I've made a video call from a SIP softphone to a PBX ACD queue that was bridged by FS. I set the PBX(AASTRA) as a gateway in FS. Here is the scenario. Softphone --> FS --> PBX --> IVR |---> Agent SIP phone The call can be established with no video. If I make the call bypass FS like this: Softphone --> PBX --> IVR |---> Agent SIP phone The video call can be make successfully. All these device are all behind NAT. I set the parameters renegotiate-codec-on-reinvite=true, inbound-late-negotiation=true and still no video. I can't figure out why the video missed. Here is the FS siptrace. http://pastebin.freeswitch.org/21879 Thanks in advance. Jimmy --- 由於 avast! 防毒軟體已開啟防護，所以此封電子郵件將不受病毒及惡意軟體的侵害。 http://www.avast.com

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detecting if user available before bridge
by ik 23 Jan '14

23 Jan '14

Hello, I need to find if a user is connected to FS currently. That is, not only registered (I found how to detect that<https://wiki.freeswitch.org/wiki/Function_sofia_contact>), but for example if I can communicate with it's soft-phone. I need to know if the soft-phone crashed for example, and if not, then not to send the bridge command at all, but actually been able to do something back in my application because of it. Is there a way to send for example an "option" header of some sort to a specific user prior to bridge command, and know it's result ? Thanks, Ido

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Wanpipe with recent versions of Linux
by Steve Underwood 23 Jan '14

23 Jan '14

Hi all, Does anyone run wanpipe with any recent version of Linux? Wanpipe 7.0.8 seems fine when compiled for Centos 6.4. However, If I try any recent kernel (e.g. the ones in recent versions if Fedora) wanpipe fails to build for a variety of reasons (i.e. if I work around one blockage I just keeping hitting another one). Regards, Steve

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cant call python script
by Joli Martinez 23 Jan '14

23 Jan '14

Hello, I have a python script located in /usr/local/freeswitch/scripts. I have a lua script in the same directory calling the python script. If I call it straight lua luascript.lua it works. It calls the python script and returns the correct results. When I have FS call the srcipt I get an error python file not found. Any ideas?

1 0

Attended transfer using bridge???
by Ali Pey 22 Jan '14

22 Jan '14

Hello, We need to implement attended transfer and cannot use att_xfer since it does not have all the features we would like to have. Let's say we are in a bridged call. The b-leg presses a *digit key and goes to the transfer menu. At this point, 1- The a-leg should listen to music and I do this now with playback 2- The b-leg calls out using bridge After the 3rd party answers that call, there is no speech path. Has anyone implemented transfer using bridge? This is for trunk calls that there is no transfer key - I know that sip phones have transfer feature. Any guidance or suggestion on how it's best to implement such a feature? Regards, Ali Pey

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Don¹t Forget the Weekly Call... This Week Brian West and FreeSWITCH NAT plus news and notes
by Ken Rice 22 Jan '14

22 Jan '14

Call in to hear Brian with a brief NAT overview! How to handle NAT with FreeSWITCH When 1PM EST at sip:888@conference.freeswitch.org or http://fs0.us/call888 Don¹t forget to follow us on Twitter, Facebook or your favorite social networking sites! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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try to build rpms git master with legacy/mod_lua
by Karsten Horsmann 22 Jan '14

22 Jan '14

Hi List, in commit 2643ac2b96c891b0827391a506f44c6cfea84378 Anthony added the old 5.1 mod_lua again. I want to change the freeswitch.spec to get this legacy/mod_lua build too as rpms (on CentOS 6.5). Did someone do that changes to freeswitch.spec already? My 4 steps to build are these (using the ci helperscripts). git clone git://git.freeswitch.org/freeswitch cd freeswitch export version_number="1.5.8b.df193ac" export build_number="2" ./scripts/ci/src_tarball.sh ${version_number} ./scripts/ci/get_extra_sources.sh ${version_number} ./scripts/ci/rpmbuilder.sh ${version_number} ${build_number} Thanks in advance. -- Kind Regards *Karsten Horsmann*

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