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Re: [Freeswitch-users] dtmf via info
by Meftah Tayeb 18 Feb '11

18 Feb '11

ok, 1. why you need sip info DTMF? is not standard 2. you could try out <action application="set" data="dtmf-type=info"/>. if don't work, set it in the profile. 3. i strongly recomand you to use rfc2833 or inband dtmf. thank you Le 18/02/2011 09:57, Nikolay Kondratyev a écrit : > > Hi all, > > I’d like to send dtmf via info and I’d like to set it in the dial plan. > > Do I just need to set dtmf_type channel variable in the dial plan? > > I tried to set<action application="set" data="dtmf_type=info"/> in the dialplan, but I do not see Info messages…. > Should it work in this way at all? > Did I miss something? > Thanks in advance, > Nikolay. > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is > believed to be clean. > > > _______________________________________________ > 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 > -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

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which language to implement dial plans in?
by Edward de Jong 18 Feb '11

18 Feb '11

I am new to freeswitch, and wondering which of the many languages to implement my rather complex dial plan in. I know i can't express my desires in the normal XML language, because I need looping and database access, so which language is the best tested and most reliable in terms of usage with FreeSwitch. My finalists so far are Lua, Python, and good old C. Haven't used python or lua yet, but I figure C will have a big performance advantage. but maybe not important considering how much greater the CPU load in the switch is devoted to handling audio streams... edj

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dtmf via info
by Nikolay Kondratyev 18 Feb '11

18 Feb '11

Hi all, I’d like to send dtmf via info and I’d like to set it in the dial plan. Do I just need to set dtmf_type channel variable in the dial plan? I tried to set <action application="set" data="dtmf_type=info"/> in the dialplan, but I do not see Info messages…. Should it work in this way at all? Did I miss something? Thanks in advance, Nikolay. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Call Recording
by Jerry Richards 18 Feb '11

18 Feb '11

Does the *2 DTMF method still work for recording calls? I recall this working before without needing to make changes to the dialplan XML, but right now it does not appear to work on the Sep 20 Freeswitch trunk. Thanks, Jerry

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IVR application
by Jason b.a 18 Feb '11

18 Feb '11

Here is a call flow sample for playing a prompt, later on i need to make my IVR application virtual using cloud computing , in this example freeswitch can play the role of media server, for media processing , and the application server for signal processing and controlling the media server sure, is it right ! or should i use different media server such as SEMS ! sip Caller Application Server Media server -----------invite----------------> ----------------invite---------------------> <---------Trying----------------- <--------------200ok---------------------- <-------------200ok------------- --------------Ack---------------> -----------------Ack-----------------------> Caller in interactiong with media server ---------Info(play prompt X.wav)---------> <----------------Ok---------------------------

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IVR application
by Jason b.a 18 Feb '11

18 Feb '11

no i need to host my ivr on a application server, but how can i control a running FreeSWITCH server from another machine, for example how the application will send an info message to freeswitch to play a prompt !

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Unable to dial IVR menu options
by Charles Bujold 17 Feb '11

17 Feb '11

Here is the scenario, I have a telephone connected to a Grandstream HT503 fxs with a pstn connected to the fxo and a SIP connection via the Lan to a freeswitch PBX. If I dial my voicemail (located on the SIP Freeswitch) I get the IVR for the voice mail but when I dial my password it does not get the numbers (password#) I dial to hear my messages. If I try this from a SIP phone everything works and I can dial an IVR option. It looks like if Freeswitch is not able to pickup the dial numbers of the phone once the call is started. Any suggestion on how to fix? Thanks cjb

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Re: [Freeswitch-users] txfax and multiple files
by Madovsky 17 Feb '11

17 Feb '11

I didn't succeed so I use tiffcp to condensate all tiff files in one with several pages ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Thursday, February 17, 2011 11:54 AM Subject: Re: [Freeswitch-users] txfax and multiple files not yet, will do test later today. If I remember it's possible by separate every path files by a space or comma Will confirm soon in case of it's not possible, can I run tfax app several times while the channel is up ? Thanks ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Thursday, February 17, 2011 11:41 AM Subject: Re: [Freeswitch-users] txfax and multiple files Did you figure this out? I didn't know that you could queue up multiple files for transmission, but if you can then I'd like to know so that I can get it properly documented... -MC On Wed, Feb 16, 2011 at 12:19 AM, Madovsky <infos(a)madovsky.org> wrote: What is the right syntax to send multiple files with mod_spandsp txfax ? thanks _______________________________________________ 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 ---------------------------------------------------------------------------- _______________________________________________ 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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TLS / SSL improvements patch request for testing
by Mitch Capper 17 Feb '11

17 Feb '11

Earlier today on trunk there was a fix committed so that if you have had an issue getting registrations to work over TLS/TCP try again as it may just work now without any config changes. In addition there is a set of patches to expose some additional TLS options. The patches are at: http://jira.freeswitch.org/browse/FS-3071 the first patch (sofia_tls_extra_options.patch) is the main one, the other two are only needed if you want to use private keys that are passworded, note the upstream patch needs to be applied in the freeswitch/libs/sofia-sip. If you are using freeswitch to connect to a remote server securely it is highly advisable to be using the options in this patch. Right now freeswitch does no certificate validation which means you could easily be MITM defeating the encryption. The main config options for sip profiles are: tls-verify-policy - this is the important one, set to 'out' to ensure remote servers we connect to(gateways) have a valid certificate, set to 'in' to ensure any clients connecting to us have a valid certificate, and set to 'all' to validate both. tls-only - set to true and freeswitch will not even bind for un-encrypted registrations tls-no-verify-date - set to true to disable checking of the dates in certificates tls-verify-depth - max depth for validating a certificate and optionally with the extra two patches: tls-passphrase - set the passphrase for the private keys used NOTE: once you enable certificate validation you will want to ensure in your tls-cert-dir you have a cafile.pem which contains the entire certificate chain to validate connections. Most of the time this is going to just be the CA cert for the certificate issuer. Also you can optionally put an agent.pem in the cert-dir this is the key/cert that will be used for freeswitch when it connects out (its client certificate/key). It is not needed however if the remote side is not validating you. If you are running into problems try turning up the debug level by setting "sofia loglevel tport 9" it should give you more details about why things fail (as otherwise you may just see gateway timeout messages or something similar). Please test if you can! Also for those not using TLS /SSL right now and going over the internet you should try and switch to it if possible. There is very low overhead (using sslv23 you still have udp packets for the actual rtp stream and they are just 167 bytes vs 164 without encryption) to get started with it take a look at: http://wiki.freeswitch.org/wiki/SIP_TLS Note I would suggest sslv23 over tlsv1, sslv23 keeps the RTP stream as UDP which should mean least chance of a difference between no encryption and with encryption. The page does not take into account these new changes yet however. For a 30 second tutorial on getting started in linux: from the freeswitch bin dir on the server: ./gentls_cert setup -cn pbx.yourdomain.com -alt DNS:pbx.yourdomain.com -org pbx.yourdomain.com ./gentls_cert create -cn pbx.yourdomain.com -alt DNS:pbx.yourdomain.com -org pbx.yourdomain.com then make sure your freeswitch user owns the conf/ssl folder generated. Copy cafile.pem from it to any clients then set the following options in your profile config: <param name="tls" value="true"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-version" value="sslv23"/> <param name="tls-verify-policy" value="none"/> and you should be able to have clients register securely( again see wiki page for more information). For freeswitch client: enable the above tls options, put the cafile.pem from the server in the conf/ssl folder, set: <param name="tls-verify-policy" value="out"/> and in the gateway: <param name="register-transport" value="tls" /> <param name="contact-params" value="tport=tcp"/> <param name="realm" value="sip.yourserver.com:5061" /> (or whatever the TLS port you are using is) ~Mitch

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Parallel compilation on a Linux machine
by mazilo 17 Feb '11

17 Feb '11

I tried to use make -j 4 to compile FS git on my AMD64 PhenomII X3 machine. The compilation doesn't seem to fully utilize all the 3 CPU cores. As a matter of fact, it takes almost the same amount of time (about 55 minutes) to compile FS git with only -j 4 switch. So, does anyone here know how to make compile FS git in parallel to make use of all CPU cores? This way, I don't have to spend about an hour just to compile FS git. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Parallel-compilation-on-a-Lin… Sent from the freeswitch-users mailing list archive at Nabble.com.

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