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Re: [Freeswitch-users] FreeTDM configuration
by ashish gautam 14 Mar '13

14 Mar '13

Hi Michael, I get this error " 2013-03-14 12:33:19.476237 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [DESTINATION_OUT_OF_ORDER] 2013-03-14 12:33:19.476237 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] " On Thu, Mar 14, 2013 at 12:19 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > 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..." > > Today's Topics: > > 1. FreeTDM configuration (ashish gautam) > 2. Re: SaltStack + deployment techniques (Gabriel Gunderson) > 3. Re: Recording channels input and output voice in two separate > files. (Michael Collins) > 4. Re: FreeTDM configuration (Michael Collins) > 5. Re: SaltStack + deployment techniques (Henry Huang) > 6. Re: Recording channels input and output voice in two separate > files. (Nikhitha) > > > ---------- Forwarded message ---------- > From: ashish gautam <gautamashish09(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Thu, 14 Mar 2013 10:47:06 +0530 > Subject: [Freeswitch-users] FreeTDM configuration > 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) > > > > ---------- Forwarded message ---------- > From: Gabriel Gunderson <gabe(a)gundy.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 23:47:58 -0600 > Subject: Re: [Freeswitch-users] SaltStack + deployment techniques > On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] > <cal.leeming(a)simplicitymedialtd.co.uk> wrote: > > I'm still not entirely happy with the overall procedure and always > looking > > for new/better ways to improve it.. but SaltStack is a clear winner, > leaving > > puppet/chef in it's dust. > > Sounds like you're getting close to being happy :) > > When we do some Salt Modules and States specific to OpenSIP and > FreeSWITCH, we'll be sure to open source them. > > Happy hacking! > > > Gabe > > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 22:50:42 -0700 > Subject: Re: [Freeswitch-users] Recording channels input and output voice > in two separate files. > You can use an api_hangup_hook to call system and do two sox commands: > > sox infile.wav outfile.left.wav remix 1 > sox infile.wav outfile.right.wav remix 2 > > -MC > > On Wed, Mar 13, 2013 at 9:52 PM, Nikhitha <nikhitha.voxta(a)gmail.com>wrote: > >> Thanks for the reply... >> In the path we are specifying the path where the recorded file should be >> stored, here we are only specifying a single wav file path then how two >> files will get recorded one as inbound and one as outbound?? >> >> >> >> -- >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 22:54:10 -0700 > Subject: Re: [Freeswitch-users] FreeTDM configuration > You need to look at the log output when mod_freetdm is being loaded. There > are several ways to do that, but the quickest is probably just to rotate > log files, restart freeswitch, then rotate log files again. You'll have a > file like: > freeswitch.log.2013-03-14-10-45-30.1 > > Look in there for errors when mod_freetdm is loading. Sometimes those > errors point out the obvious, like it can't find the card or the DAHDI > drivers aren't loading/responding, etc. > > -MC > > On Wed, Mar 13, 2013 at 10:17 PM, ashish gautam <gautamashish09(a)gmail.com>wrote: > >> 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) >> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Henry Huang <red.rain.seven(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 23:36:27 -0700 > Subject: Re: [Freeswitch-users] SaltStack + deployment techniques > It's the first time I hear about SaltStack. Can you briefly explain why is > it better than Puppet or Chef. I would pick Chef out of the 2 because it's > using Ruby natively and is being adapted by Amazon AWS. So if you ever need > some kind of hybrid architecture to run production or development servers > on AWS, you will spend minimal effort for deploying those given that you > can reuse your Chef cookbooks. > > Henry > > > On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson <gabe(a)gundy.org>wrote: > >> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >> <cal.leeming(a)simplicitymedialtd.co.uk> wrote: >> > I'm still not entirely happy with the overall procedure and always >> looking >> > for new/better ways to improve it.. but SaltStack is a clear winner, >> leaving >> > puppet/chef in it's dust. >> >> Sounds like you're getting close to being happy :) >> >> When we do some Salt Modules and States specific to OpenSIP and >> FreeSWITCH, we'll be sure to open source them. >> >> Happy hacking! >> >> >> Gabe >> >> _________________________________________________________________________ >> 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 >> > > > > ---------- Forwarded message ---------- > From: Nikhitha <nikhitha.voxta(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) > Subject: Re: [Freeswitch-users] Recording channels input and output voice > in two separate files. > 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. > > > > _______________________________________________ > 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 > > -- REGARDS ============================================ *Ashish Gautam* (+918802865008)

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Advice on scalable design pattern
by Cal Leeming [Simplicity Media Ltd] 14 Mar '13

14 Mar '13

Hello all, I'm currently looking at the various different ways you can deploy FreeSWITCH in a scalable manner, but struggle a little bit on design. The sweet spot I'm trying to find is one where I can scale out capacity by simply throwing more servers at it. In an ideal world, this would mean support for; * Have multiple users from multiple domains to be spread over multiple servers... a single domain should not be restricted to a single FreeSWITCH instance * Have no single point of failure within the structure * Have no single point of bottleneck within the structure * Should not use OpenSIPS.. (I suspect this might get me a lot of flack, but seriously, I'd rather write my own in python or ZXTM traffic script than use OpenSIPS lol). So far, the best option I can come up with is (although I'm not sure if it's the best available); * Proxy sitting in front of all backend FreeSWITCH instances, acting in a media proxy fashion only (dual pair of proxies in active/passive mode) * Proxy tracks registrations to the appropriate backend instance, and makes their session sticky * If backend instance needs to make a call to another user in the same domain, it bridges to the call to back to the proxy, the proxy then determines which other FreeSWITCH instance has the user then routes the request accordingly. If the call is to an external destination, the proxy will route it to the traffic aggregation switches (which is basically another pair of FreeSWITCH instances), which then gets routed to the upstream provider.. this means you only have to maintain 2 sets of trunk configuration.. so when you need to scale out your freeswitch backends, it doesn't require putting in a request to your upstream providers for an additional set of trunks. * The bottleneck within these clusters is the dual proxies in active/passive mode.. you could fix this by allocating customers to a specific cluster (rather than instance), thus controlling which customers go to which proxy.. if an entire cluster dies, you can re-route that clusters traffic to a different cluster. The other simpler option is to allocate domains to a specific backend instance.. but this really doesn't feel clean.. it means a customer cannot scale past the floor limit of a single instance, it has less redundancy, and overall just feels wrong. Any general thoughts/comments on this would be much appreciated. Thanks Cal

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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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