- FreeSWITCH-users - FreeSWITCH Mailing Lists

Accepting google talk friend requests
by Tapan Parikh 21 Aug '09

21 Aug '09

Hi Folks - Im a relative newbie freeswitch and first of all wanted to thank you for all the great work u have done here. My question is about mod dingaling, and specifically being able to get incoming calls from Google Talk. Ive got the client set up on Freeswitch, and am able to receive calls, IMs, etc. once I accept the friend invite. However, new accounts cannot talk to FS until they are already friends. Is there a way to get FS / mod dingaling to automatically accept incoming friend requests? I saw something in the archives about this, but no obvious resolution. People seem to say this is already true, but I dont even see the incoming invite requests coming in w/ DL in debug mode. Thanks in advance, Tapan

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Dialing SIP URL issue
by Henry Huang 21 Aug '09

21 Aug '09

Hi: I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is : *X-lite(softphone) dials -> 1009(a)4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it) * Please see pastebin for full log. http://pastebin.freeswitch.org/10089 ignore anything after line 80, because it's not my point, and the destination is a fake address. I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume there must be a way of passing full sip url to the dialplan. Here is the example dialplan expecting sofia to pass it a full sip url:  <extension name="sip_uri"> <condition field="destination_number" expression="^sip:(.*)$"> <action application="bridge" data="sofia/${use_profile}/$1"/> </condition> </extension> Thanks -- Henry Huang

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Re: [Freeswitch-users] Better results from mod_vmd
by Eric des Courtis 21 Aug '09

21 Aug '09

Matt, Okay the good news is vmd should be able to handle these cases. The bad news is for whatever reason they are not getting detected at the moment. vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help. The following seem to use the same beep: vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise You can try to play with these values: POINTS 32 VALID 22 MAX_CHIRP 22 If that doesn't work let me know I will try to improve the algorithm to detect the providers. Cheers! Eric des Courtis On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong(a)gmail.com> wrote: > Hi Eric, > Thanks for the response. I had tried emailing you @brenbria.com and the > email had bounced, thanks for responding to my mail. > If you'd be interested I .zipped up my sample voicemail beeps > at http://bandcon.hellohunter.com/vmd_wav.zip > I'm relatively new to telephony, but can you point me in the right direction > for figuring out if the beeps are sinewaves. About as far as I've come with > audio is being able to open the .wav files in audacity. Any website > recommendations I can read? Thanks so much. > --matt > > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis > <eric.des.courtis(a)gmail.com> wrote: >> >> Matt, >> >> You must first capture the audio beeps and verify that they are sine >> waves. If not, simply tweaking the algorithm will not give you better >> results. >> >> It might be possible to use FFT and I would be happy to help you >> implement such a solution but keep in mind FFT is very very demanding >> on the hardware. Ideally what you want to find out is what functions >> was use to generate the beep in the first place so that it can be >> detected. Is it two sines waves like in DTMF? Or something more >> complex? >> >> Anyway my email is eric.des.courtis(a)benbria.ca. >> >> Cheers. >> >> Eric des Courtis >> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong(a)gmail.com> wrote: >> > I tried emailed Eric, seeking advice on this, but his email (the one in >> > the >> > source code) is bouncing email (invalid user), so thought I would ask >> > here >> > instead. If anyone has eric's new email address, I'd be interesting in >> > it. >> > >> > I did some tests with mod_vmd this afternoon, but I'm only finding about >> > 33% >> > of the voice mail beeps and did have 1 false-positive in my test of >> > 7 voice >> > mail machines. I've recorded the audio of the session in .wav files that >> > were both successful and not, as a comparison. I can upload the .wav >> > files >> > if they would be useful. >> > mod_vmd works great for voicemails of Skype Users, and kall8.com, but >> > has >> > issues dealing with mobile phone carriers. >> > sprint - not successful >> > tmobile - not successful >> > verizon - not successful >> > panasonic home answering machine system - not successful >> > kall8 - SUCCESS >> > skype - SUCCESS >> > I'm wondering if you can recommend a simple fix, like changing some of >> > the >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source >> > file, or if better success requires more complex analysis. Do you have >> > any >> > recommendations on how this might be done? Listening to the .wav's >> > its apparent the beeps are not as loud for the mobile phone carriers as >> > they >> > are with skype and kall8. Any guidance would be greatly appreciated. >> > --matt >> > hello hunter >> > http://www.hellohunter.com >> > voice broadcasting & hosted dialer >> > _______________________________________________ >> > 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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Is it possible to unbind the * key?
by Alan Chandler 20 Aug '09

20 Aug '09

In a previous e-mail to this list I explained some problems I was having with the re-entering a conference and the caller keys not working. Nobody responded (See "Stuck on my first attempt at dialplanning" above). I have made some good progress in getting things to work. What appears to be the case is that as I callout from the conference the first time, I call bind_meta_app in order to enable me to get back to the conference from the callout call. When I get back to the conference the * key no longer works as a conference control, presumably because bind_meta_app still has control of it. Is there a way to unbind it before re-entering the conference? -- Alan Chandler http://www.chandlerfamily.org.uk

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send sip options message
by Christian Löschenkohl 20 Aug '09

20 Aug '09

hi does anybody know how to send a sip options message to a registered user, using the event socket or something else build in freeswitch i think the ping parameter does something like this for gateways. what i want/need is the same thing that is provided in asterisk with the qualifying option, to see how "reachable" a certain client is. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Sharing Presence Information Across Separate Offices
by Kenneth Shaw 20 Aug '09

20 Aug '09

I apologize in advance if this message duplicates a previous post or the topic has been covered elsewhere. If that's the case, a pointer would be appreciated! I have multiple offices each running Freeswitch and a number of Polycom phones and softphones. I would like to share the presence information between the offices, such that if a person in Office B can see if a person in Office A is on the phone. I personally don't know of any way to do this through Freeswitch as it stands. I have thought about using a VPN for the offices and just setting up the phones to all register to a single server, however I don't want calls between phones in the same office (intra-office) to have to be routed all the way through a different office, which may be on the other side of the world and costs bandwidth. Any suggestions as to the best way to accomplish this would be greatly appreciated. Thanks! -- Kenneth Shaw ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949.650.4600 fax: 949.642.6044 ken(a)expitrans.com

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FS and SS7
by Dome Charoenyost 20 Aug '09

20 Aug '09

Dear All, my telco can provider SS7 (Now i use ISDN PRI). I have question about SS7. 1. Can i setup SS7 with out ss7box ? 2. I found some tutorial about libss7 in Asterisk. Is FS support all feature like that. Best Regards. Dome C.

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How to delay IVR answer during an outbound call
by Paul Li 20 Aug '09

20 Aug '09

I have a dummy question. Say, you have an outbound call to the demo IVR as below: originate sofia/gateway/myvoip/19876543210 5000 How do I delay the IVR response until the recipient at 19876543210 picks up the call? I tried "ignore_early_media=true", which had no effect. Many thanks in advance.

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Re: [Freeswitch-users] "mute" channel programmatically with mod_event_socket
by Rudolf Denert 20 Aug '09

20 Aug '09

You can use the caller controlls in the conference.conf.xml to implement your own features something like mute or kick. Or do you want to mute mute other conference members like a moderator can do this. BR ----- Ursprüngliche Mail ----- Von: "Artem Shiyanov" <shiyanov(a)gmail.com> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: [Freeswitch-users] "mute" channel programmatically with mod_event_socket Hello all! I'm trying to implement "mute" feature with mod_event_socket: I want programmatically mute/unmute some channel in a call.. And I don't see any other ways except to use conference room with special rule "mute". Anybody knows the better way? Thanks, Artem _______________________________________________ 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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Recommendations for an interface between FreeSwitch and ANSI C code
by Matt Riddell 20 Aug '09

20 Aug '09

Hi, We have an application written in ANSI C which currently talks to the Asterisk Manager to make phone calls. We're possibly looking at converting this to FreeSwitch. At the moment it has an abstraction layer from Asterisk and speaks to between 1 and 80 Asterisk machines using round robin for distribution. Would you recommend that I use the mod_event socket and basically work with FreeSwitch in the same way as I work with Asterisk or am I overlooking a possibly different way of doing things? Cheers, Matt Riddell

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