- FreeSWITCH-users - FreeSWITCH Mailing Lists

IVR: Techniques for logging DTMFs to CDR
by Michael Collins 29 May '08

29 May '08

Gang, I'm working on my outbound IVR and I'm wondering if you have any thoughts on this topic: I'd like to make a record of the digits dialed by the called party. For example, I have an IVR menu that says, "to repeat these options, please press star." I'd like to record how many times the called party pressed star, along with all of the other digits. What do you suggest? One idea is to create a custom channel variable, like "digits_recd" and make it a string of digits, appending a new digit to the string with each key press on the part of the called party. I do like this idea because it preserves the order in which the digits are received. Any other suggestions? I'd like to know what you've done. BTW I'm not using any scripting - this is all being done with the XML dialplan, phrase macros and ivr menus. Thanks, Michael

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Re: [Freeswitch-users] RESTful Bounty ..... Putting the competition to REST with RestFul VoIP2.x services
by Ward Mundy 29 May '08

29 May '08

> On Tue, May 27, 2008, Michael Collins wrote: > > > Re: Any chance we'll see some of your sweet recipes on nerdvittles with a little FS baked in? :D > You bet. In fact, we're raising a bounty on the trixbox forums to port FreePBX to FreeSwitch as we speak... http://trixbox.org/forums/trixbox-forums/open-discussion/freeswitch-1-0-0-p… Raised $3,000 in the first afternoon!

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Conditions in the dialplan
by Jonathan K. Creasy 29 May '08

29 May '08

I've been looking around a little bit and I haven't seen anything on this yet so I wanted to ask here. I want to have a dialplan that captures certain SIP responses and takes different actions based on the response received. For example, if I place a call out to a peer and get back "486 Busy here" I want to do "action1". If I get "302 Temporarily Unavailable" I want to execute "action2". What's the best way to go about this? -Jonathan No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.24.1/1469 - Release Date: 5/27/2008 1:25 PM

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SRTP in PhonerLite and Freeswitch
by Krzysiek 28 May '08

28 May '08

Hi I have 2 softphones PhonerLite (they support SRTP via SDES ) and the freeswitch (windows RC1 version) server and I wanted to make secure call between those two endpoints (SRTP). I spend whole day on testing this scenario and my conclusions are: - when the option: <action application="export" data="sip_secure_media=true"/> is uncommented, and both enpoints have enabled SRTP then: 1) Initiator of the session sends SIP Invite with a=crypto paramter and supported codecs 2) Freeswitch receives SIP Invite and sends SIP Invite to the receiver (also with the crypto) 3) Receiver receives the SIP Invite with the a=crypto parameter and he sends back supported codecs with 200 OK message (but without a=crypto parametr. Is that ok? I'm afraid not) 4) Freeswitch sends 200 OK message but witout any codecs: m=audio 0 RTP/AVP 19 and no a= parameters! 5) Final result is that the second leg of the session between Freeswitch and receiver has SRTP transport enbaled and the first leg (initiator- Freeswitch) doesn't hear anything - no codecs! However Freeswitch is sending RTP (not SRTP) pacekets to the initiator. Could someone explain to me, what is going on, and why freeswitch doesn't forward codecs accepted by the receiver to the initiator? Is it a PhonerLite's bug or freeswitch's? Maybe someone has tested SRTP with the PhonerLite softphone or any other free softphone with srtp support? When I uncommented: <param name="Inbound-no-media" value="true"> everything works fine. The parameter <action application="export" data="sip_secure_media=true"/> doesn't change anything then (but i cound miss something). Thanks for help Chris

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SS7 support in FS
by Alex Kinch 28 May '08

28 May '08

Hi all, A colleague of mine who currently uses Asterisk was asking me the other day about SS7 support in FreeSWITCH - and whether people have such a thing in production. I don't think he's looking for MAP support, which makes life a little easier. I know of at least one company that's used FS with TelcoBridges hardware to achieve an SS7 interconnect, but thought I'd throw the question out to the list for discussion. Thanks, Alex

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SS7 and SIP
by imthiyaz＠peopletech.co.in 28 May '08

28 May '08

HI We want to try generate 5000 simultanious Voice broadcast calls . can the below config will work? SS7 Links< ----> Sangoma SMG<------->FreeSwitch<-------->Broadcasting Application ( SIP based) Thank you Imthiyaz -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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MySQL support in FS?
by Aadilkhan Maniyar 28 May '08

28 May '08

Hi All, Does FreeSwitch support the MySQL database? I could not find any documentation related to the same in the FS wiki. If FS does not support MySQL or unixODBC how are SIP registrations made persistent? Regards, Aadil

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Help Creating an Inbound Profile for DIDWW
by Klaus Teller 28 May '08

28 May '08

Hi Folks, I have 30$ bounty for the following task. I want a FS profile to support DIDWW. I intend to use Les.net for outbound calls and DIDWW for incoming. Right now, the outbound part seems to be OK. But I'm having difficulties creating a profile (and the related diaplan) for receiving calls. There is no username/password provided by DIDWW. In the diaplan, I want just one extension (something like extension 9998). Anyone interested? Thanks, Klaus. -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser

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need billing
by AJ Singh 28 May '08

28 May '08

need billing for freeswitch aj(a)btel.ca On 5/27/08, imthiyaz(a)peopletech.co.in <imthiyaz(a)peopletech.co.in> wrote: > > > HI > > We want to try generate 5000 simultanious Voice broadcast calls . > can the below config will work? > > SS7 Links< ----> Sangoma SMG<------->FreeSwitch<-------->Broadcasting > Application ( SIP based) > > Thank you > Imthiyaz > > -------------------------------------------------------------------- > mail2web.com – What can On Demand Business Solutions do for you? > http://link.mail2web.com/Business/SharePoint > > > > _______________________________________________ > 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 > -- AJ Singh CTO http://www.btel.ca Phone1: +1 647-258-0685 Phone2: +1 780-701-7311 efax: +1 (888) 205-0087

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Meeting today.. 1PM Central Time.
by Brian West 28 May '08

28 May '08

http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_28 Please add anything to this list you think we need to talk about. As usual we'll breeze thru jira, docs and various other things that come up along the way. Please remember to DIGG the 1.0 release article at http://digg.com/software/FreeSwitch_1_0_Released Brian West sip:brian@freeswitch.org

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