March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Announcement: FreeSWITCH 1.2.7 Is Now Available - Centos rpms?
by Karsten Horsmann 14 Mar '13

14 Mar '13

Hi all, thanks for the great work. Will someone update the files.freeswitch.org/yum repo with new 1.2.7 rpms for CentOS? Kind Regards Karsten

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Freeswitch not falling back to digest auth
by Garey Arrington 14 Mar '13

14 Mar '13

First of all let me say I am new to freeswitch. I am working with a bluebox server and I am having problems with registrations and a few other things. First thing I need to figure out is why I can't get freeswitch to user digest authentication. When attempting to register I get the following: 2013-03-14 07:39:19.886136 [WARNING] sofia_reg.c:1940 IP xxx.xxx.xxx.xxx Rejected by register acl "net_list_5" The phone will not register. Here is my acl.conf.xml: <?xml version="1.0"?> <include> <configuration name="acl.conf"> <network-lists> <list name="net_list_1" default="deny"/> <list name="net_list_2" default="deny"/> <list name="net_list_3" default="deny"/> <list name="net_list_4" default="deny"/> <list name="net_list_5" default="allow"/> <list name="net_list_6" default="allow"> <node bluebox="net_list_item_1" type="allow" cidr="0.0.0.0/0"/> </list> <list name="net_list_7" default="deny"> <node bluebox="net_list_item_2" type="allow" cidr="173.208.83.50/32"/> </list> </network-lists> </configuration> </include> I really need some guidance here, I am not sure where to go from here. I can't seem to find this particular issue documented anywhere.

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Re: [Freeswitch-users] Recording channels input and output voice in two separate files.
by Clive Lansink 14 Mar '13

14 Mar '13

I don't think so unless someone updates the record_session application. But in the meantime, sox would be a good way to split the recorded stereo file into two separate files if that's what you need. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Nikhitha <nikhitha.voxta(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording channels input and output voice in two separate files. Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) Without using sox ,do we have any thing that we can directly do with freeswitch to record them in seperate files? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-… Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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uuid_displace + uuid_fileman
by Tamas.Cseke 14 Mar '13

14 Mar '13

Hello, We'd like to play a file to both legs of the call, while they still hear eachother I tried uuid_broadcast, the playback can be managed (pause/seek) with uuid_fileman but the speech cannot be heard with uuid_displace the playback can be mixed to the speech, but it can't be controlled As far as I understand playback can be controlled, but it absorbs the original speech displace uses media bugs, that can't be controlled (all of them could be paused on the channel) Is it a way to mix frames and pause the playback as well? Is it possible to add an option to playback to "skip frame absorbing" or add api to pause the file hande of displace media bug? Thanks in advance, Tamas

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Starting recording
by Usama Zaidi 14 Mar '13

14 Mar '13

Hi, A hacky way to do it would be to place the user in a conf bridge, have a local channel dial to your IVR, monitor the speaking, stop speaking events on the UUID and control the playback of audio. -Regards. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Thursday, March 14, 2013 2:30 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 142 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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Starting recording
by Nikhitha 14 Mar '13

14 Mar '13

Suppose if we take a customer service.In this if the system is playing a message then user is waiting to give some input through voice.Here in this service the user can give the input and it gets recorded and proceed to further while system is playing the message(At that time the message that system is playing may be skipped or interrupted there itself and take the user recording). So in freeswitch do we have an option like this,can we start recording while system is playing a message?So that the message that is playing to be stopped and whatever the user is recording should get recorded.(According to my knowledge these are two applications playback & record these will be executed one by one.Is that possible to run two applications at the same time?) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Starting-recording-tp7588614.… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Recording channels input and output voice in two separate files.
by Usama Zaidi 14 Mar '13

14 Mar '13

Hi, If you have record_stereo = true (its true by default), the resultant file using uuid_record will have be stereo, one channel containing inbound audio and the other containing outbound. -Regards -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, March 13, 2013 8:50 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 81, Issue 131 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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FreeTDM configuration
by ashish gautam 14 Mar '13

14 Mar '13

Hi, I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it does not show any output. I guess the channels have not been configured properly. I am not able to figure out what is happening. Please help me out. I am using a Digium single span E1 card. Dahdi and libpri have been properly installed and are working fine. -- REGARDS ============================================ *Ashish Gautam* (+918802865008)

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switch loading
by shayne.alone＠gmail.com 14 Mar '13

14 Mar '13

Hi all I had planned to make a max load test on my FS switch: we send in coming calls to opensips and balance them between to FS, then terminate them on one other asterisk (with just MoH app). first on 1K calls ( which will load 1K -2leg- sesssion on each of fs), we found the "max-proceeding" limit that control the open sip dialogs... is there any other limit or control witch I need to look forward? or have affect on hi loads? ** I don't think that 1K sessions are really more.... -- Regards, Ali R. Taleghani

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SSL/SRTP VS SSL/ZRTP
by mehroz 13 Mar '13

13 Mar '13

Hi all While configuring FS with secure setup which includes SIP secure over SSL/TLS and RTP with ZRTP, yet i am still unable to get it done. I came across a question i.e, how do we tell FS to do Secure SIP signalling (SSL/TLS) with securing RTP as ZRTP (not SRTP). I am very confused as what comes first! ? If I have enabled ZRTP settings in profile and vars.xml, and if i configured SSL in SIP signalling, how to tell it to not to secure RTP with SRTP, but ZRTP. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-SRTP-VS-SSL-ZRTP-tp758821… Sent from the freeswitch-users mailing list archive at Nabble.com.

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