June 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Key System Emulation
by Matt Darnell 27 Jun '08

27 Jun '08

It seems like Freeswitch is a real player in the PC based PBX space. I think the first product in this space that can emulate a key system will take over the space in very short time. Metaswitch, Broadsoft, etc have shared line appearances but none of the smaller systems have it. It must be hard to do, or everyone would be doing it. If Freeswitch could offer the kind of functionality it would take over: A call would come in, via SIP, Zaptel, etc. User answers the call User puts it on hold That line appearance blinks on all the phones Any user presses the button to get the call It would be OK if it only worked with Polycom, or Linksys, or Snom,etc...as long as it worked _every_ time. -Matt

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Java Support
by Klaus Teller 27 Jun '08

27 Jun '08

Hi Folks, Questions regading Java (not Javascript): 1) How do I originate a call? I tried somethis as following without success: JavaSession session2= new JavaSession(); session2.originate(null,"sofia/gateway/did.voip.les.net/14163442000"); The error messgae I get is that session2 is not initialized. Any idea? 2) Assuming I have two legs session1 and session2. Bridging them would be done with freeswitch.birdge(session1, session2) is that correct? I coudln't test this since I'm not able to create session2. Quick Observations: 1) The wiki says that Java 1.6 would cause FS to crash. But, actually the current Java 1.6 works just fine while the current Java 1.5 (update 15) causes the FS to crash. 2) Whatever path to Java is specified when building FS, when starting it, it will look for Java at: /usr/java/jdk1.6.0/jre. Looking forward to your feedback. Klaus L'Imbecile. -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser

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Re: [Freeswitch-users] config / scripting questions from voice mail integration attempt
by John Wehle 26 Jun '08

26 Jun '08

> if you do > session.execute("transfer", "<some ext>"); > and exit the script. > > That goes back to the dialplan for a *new* lookup so now your variables can > be used in conditions. Thanks, -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Re: [Freeswitch-users] config / scripting questions from voice mail integration attempt
by John Wehle 26 Jun '08

26 Jun '08

> you can make a variable that is the entire data portion of the voicemail > call > > <action application="javascript" data="vm.js"/> > <action application="voicemail" data="${vm_data}"/> > > then in vm.js set vm_data var to the entire args to the voicemail app > > in this setup you can assume the js will set the var and then the vm_data > will be expanded in time. Perhaps I should have supplied a different example. Sometimes the inband DTMF says the caller is checking their messages, sometimes the caller is leaving a message, sometimes the PBX just wants the message waiting lights to be refreshed, sometimes the PBX is logging a voicemail port which didn't respond, etc. I.e. the set of applications which need to be invoke varies based on the DTMF ... it sounds like the best way to handle these types of issues is to use session.execute in javascript and pay the price of leaving the javascript interpeter running. I don't see a way of just setting variables and handling the rest in the dialplan since I can't use conditionals in the dialplan after the application to parse the DTMF has been called. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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mod_shout with icecast
by e schmidbauer 26 Jun '08

26 Jun '08

has anybody successfully streamed a call to icecast using the api call record shout://user:pass@localhost/stream.mp3? if so could you let me know what OS and version of icecast you are using. thank you.

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start a conference with freeswitch
by e schmidbauer 26 Jun '08

26 Jun '08

is there anyway to start a conference automaticaly when freeswitch starts?

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control uuid
by e schmidbauer 26 Jun '08

26 Jun '08

just wondering if there is a way to create a conference with a specific uuid....im trying to have a permenent conference going with a uuid that does not change, in case the conference must be restarted i'd like the uuid to remain the same....is there anyway to accomplish this? i would appreciate any advice...thanks.

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Delay in conference
by Ivan C Myrvold 26 Jun '08

26 Jun '08

I had a conference in FreeSWITCH with a few of my friends, and one of my friends complained that when I talked, at every pause I had in my talk, the first word in my sentence was cut off. When no one is talking, there is completely slience. Is this behaviour configurable? I saw something in conference.conf.xml which is called comfort-noise, maybe that is what I should use to tune this? Ivan

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Re: [Freeswitch-users] config / scripting questions from voice mail integration attempt
by John Wehle 26 Jun '08

26 Jun '08

Anthony Minessale writes: > apiExecute is for FSAPI calls voicemail is a dialplan application. > you would want to do one of the following: > > session.execute("voicemail", "<voicemail args>"); > session.execute("transfer", "<some ext that leads to voicemail>"); Yep ... that was exactly the command I was looking for. I take it that execute is more like the C function system rather than the system call exec? I.e. control is returned to the javascript after the application finishes? > Your plan to set the variables is the best one to avoid overhead of > leaving JS running. > The only thing you can't do is set a variable in one action of your > dialplan and then expect that variable to be set to do more <condition> > tags inside that same dialplan. The inband DTMF contains information regarding whether the caller is checking their voicemail or leaving a message for someone else which means that based on the DTMF I need to call the voicemail application with different parameters. How can I have the script just set variables if I can't use those variables with <condition> tags later in the dialplan to control how to call the voicemail application? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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rtpmap + addpac
by Mihail Krivushin 26 Jun '08

26 Jun '08

I have trouble with two addpac gateways. FS: <extension name="krs-mz-addpac" continue="true"> <condition field="destination_number" expression="^39(.*)$"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/mira46_for_all/9$1(a)10.24.1.8"/> </condition> </extension> One (10.24.1.8) send SDP in reply to request from initiator: v=0 o=989138657866 1214496470 1214496470 IN IP4 10.24.1.8 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214496470 0 m=audio 23434 RTP/AVP 18 a=rtpmap:18 G729/8000/3 a=ptime:20 And initiator receive: v=0 o=989138657866 4667768711296699587 6633288917781619787 IN IP4 192.168.0.198 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214496173 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 G729/8000/3 What is it, and how can I solve this issue? -- С уважением, Кривушин Михаил Евгеньевич Системный администратор ЗАО "СОЮЗ Т" тел. +7 913 865 7 866 jid: mkrivushin(a)souzt.com icq: 218 744 127

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