May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ivr codec
by Madovsky 08 May '10

08 May '10

I just noticed in the log this : switch_ivr_play_say.c:1152 Codec Activated L16@16000hz 1 channels 20ms when voicemail is answering. But this codec is not in vars.xml list. why this codec is activated with IVR ? Thanks Franck

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Sound files French, German, and Spanish
by Diego Toro 08 May '10

08 May '10

Hi all, The lastest announcement the official release of version 1.0.6, it says, Sound files have been created for French, German, and Spanish. Where Can I get these sound files ? Thank you Diego Toro http://lacarretade.blogspot.com/

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voicemail recording problem
by babak yakhchali 08 May '10

08 May '10

Hi I'm using the below ivr menu and sending the caller to voice mail by transferring to an extension but the problem is that beep tone is also recorded! I mean the recorded voice mails begin with beep sound (deeeed)!!!! thanks <menu name="sepanta_ivr" greet-long="sepanta/mainMenu.wav" invalid-sound="sepanta/digitCollectError.wav" confirm-key="" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="1"> <entry action="menu-exec-app" digits="1" param="managed Answering 1"/>  <entry action="menu-exec-app" digits="2" param="managed Answering 2"/>  * **<entry action="menu-exec-app" digits="3" param="transfer manager-voice-mail"/> * <entry action="menu-sub" digits="4" param="sepanta_ivr_submenu"/> <extension name="manager-voice-mail"> <condition field="destination_number" expression="^manager-voice-mail$"> <action application="set" data="voicemail_greeting_number=2"/> <action application="set" data="skip_instructions=true"/> <action application="voicemail" data="default ${domain_name} kiarash"/> </condition> </extension>

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call privacy
by Meftah Tayeb 08 May '10

08 May '10

hello how do i let user do a hiden call? using privacy tools? and, if i get a hidden call and my freeswitch is a node in our telecom infrastructure, how do i force the caller id number to display? note, i am not a customer but a core switch thanks

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leg B uuid
by Madovsky 08 May '10

08 May '10

Is there an already exsting variable which represent the UUID of B leg from the A leg phone once the call is established ? (sorry Mathieu I lost all my notes of this subject yesterday) Thanks

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recording problem
by babak yakhchali 08 May '10

08 May '10

Hi I'm using the code below to dispatch customers to operators in c#. everything is fine but recording is not taking place(I've tested both the hang up after bridge false and true). thanks public class Answering : IAppPlugin { ManagedSession Session; public void Run(AppContext context) { Session = context.Session; Session.HangupFunction = hangupHook; if (. . .) //no operator available wai { Session.Answer(); if (Session.Ready()) { Session.StreamFile(m_sorryPrompt, -1); Session.SetVariable("sepanta_prompt_ended","true"); Session.StreamFile(m_musicOnHold, -1); } }//now u can talk if (Session.Ready()) { string ticket = "1234"; if (!Session.answered()) Session.Answer(); Session.Say(ticket, "en", "NUMBER", "PRONOUNCED", "FEMININE"); Session.Execute("Playback",m_readyPrompt); Session.SetVariable("exec_after_bridge_app", "record_session"); Session.SetVariable("exec_after_bridge_arg", recPath); Session.Execute("bridge",string.Format("sofia/internal/{0}%{1}", op, Session.GetVariable("domain"))); } }

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Fw: rfc2833 confusion
by Madovsky 08 May '10

08 May '10

is this the right link ? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 06, 2010 2:10 PM Subject: Re: [Freeswitch-users] rfc2833 confusion But how to catch digits form a dialplan when the call is already on ? ----- Original Message ----- From: Phillip Jones To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 06, 2010 1:43 PM Subject: Re: [Freeswitch-users] rfc2833 confusion http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle. If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on. On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos(a)madovsky.org> wrote: I anyone knows a link of a programming example of how to inject rfc2833 in RTP flow (doesn't matter the language) It would be very useful Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 06, 2010 12:39 PM Subject: Re: [Freeswitch-users] rfc2833 confusion god, so many confusion for only an audio tone ! ;) anyway if I understand it's called inband by the fact that it's in RTP flow onlly. so the rfc2833 packet is only an event payload ... ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 06, 2010 11:52 AM Subject: Re: [Freeswitch-users] rfc2833 confusion both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side. Some people refer to inband as the real tones and 2833 as not inband because it's not the tones. So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833 The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos(a)madovsky.org> wrote: HI, is rfc2833 inband or outband ? is it only event or mixed event and sound ? I'm confused after read articles on this subject Thx F _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------ _______________________________________________ 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 ------------------------------------------------------------------------------ _______________________________________________ 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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2 B Channel Transfer (2BCT) and FreeSWITCH
by Doug Albrechtsen 07 May '10

07 May '10

Is anyone having success using 2B Channel Transfer with FreeSWITCH and Sangoma? If so, what are the driver considerations and what does it take to make the transfer work? Thanks, Doug

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Registration ODBC feeded by another registrar proxy
by Rod. 07 May '10

07 May '10

Hi list, was playing with FS 1.0.6 and trying to test the registration performance of FS. (Yes I know FS is more suited as a B2BUA, but please read further :p) So I did the following: - generate one xml file with 20 000 user account like this: <include> <user id="1"> <params> <param name="password" value="1234"/> </params> </user> <user id="2"> <params> <param name="password" value="1234"/> </params> ... Then I used Sipp to test how many registration per second could be fired to the server (quad core 2.83Ghz). I setup ulimit variables, and disable nat. I got this: - using SQL Lite: unable to get higher than 80 registrations per second (in fact it's less than this number but didn't test too much this setup), I see a lot of retransmission in Sipp - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per second but not much - using ODBC and mysql: 130 reg/sec is OK With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I think it's because I'm using only one SIP profile and that SOFIA is monothreaded for this SIP profile. If I'd like to register every 60sec, the server has to support at least more than 300 registration per second. So I'm wondering if I could setup something like this: - use another SIP Proxy as a registrar and feed the ODBC "sip_registration database" of FS - FS will be able to use this database to setup a call - use FS as the outbound proxy for call routing But what about the user params that have been setup in the xml file above. I think that FS loads the user params each time a user is registered. Comments and advices are welcome. regards, rod.

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4x T1 PRI Gateway
by Max Clark 07 May '10

07 May '10

Hello, I am looking for a 4 port T1/PRI to SIP gateway to interconnect a legacy PBX to a SIP trunk. After spending some time with Google I have come up with the following list of devices that fit the required number of T1 interfaces: Dialogic DMG2120DTI Cisco AS5350 Quintum Tenor DX Patton SmartNode 4960 Audiocodes Mediant 1000/2000 Any opinions, personal experiences, advice, or additional devices I should be looking at would be greatly appreciated. Best, Max

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