January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

close channels properly
by Carole O. 07 Jan '09

07 Jan '09

Hello, I use the following code to call VoIP speakers and make an announcement: <extension name="security_announcement"> <condition field="destination_number" expression="^9001$"> <action application="set" data="conference_auto_outcall_caller_id_name=security_announcement"/> <action application="set" data="conference_auto_outcall_id_number=9001"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_announce=announcement/announcement_security_delay.wav"/> <action application="conference_set_auto_outcall" data="user/1021@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1022@$${domain}"/> <action application="conference" data="conf_speaker@profilespeaker_announcement"/> </condition> </extension> 1021 and 1022 are the speakers. At the end of the announcement, since there is no noise anymore, the speakers stop listening but they do not send any messages to tell Freeswitch it can close the opened channels. The channels are closed only after a timeout of 5 minutes. Does anybody know how I could force freeswitch to close all the channels after the announcement? I have seen there is the application sched_hangup but when I used it it only closes the channel to the caller and not the other ones. Thanks a lot for your help and Happy Holidays! Carole -- View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21127913.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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vmd
by Kristian Kielhofner 06 Jan '09

06 Jan '09

Hello everyone, I know 1.0.2 if fresh (yeah!) but what exactly is the status of vmd? The wiki page says it can't be used as an app yet it shows up in "show application" with start/stop. How might this be used from the dialplan? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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New FreeSWITCH release
by Michael Collins 06 Jan '09

06 Jan '09

FYI, If you haven't already heard, we've now released v1.0.2 of FreeSWITCH! Please digg the new release story: http://digg.com/software/FreeSWITCH_New_Release_For_The_New_Year The source can be downloaded here: http://files.freeswitch.org/ in both tar.gz and tar.bz2 formats A Windows MSI file can be downloaded here as well: http://files.freeswitch.org/freeswitch-1.0.2.msi Thanks for your support and keep on FreeSWITCHing! -MC (mercutioviz)

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BLF on Aastra 9133i
by Mark Greene 06 Jan '09

06 Jan '09

Hey guys, I have a user who is in charge of checking multiple voicemail boxes. They are registered under extension 0650, which will have voicemails as well. So the user has a total of 3 VM boxes to check. The MWI light works great for thier extension (0650), but I can't use it for the other voicemail boxes that the user is in charge of checking without asking the user to call and check each mailbox every time the MWI light comes on. The Aastra 9133i has 7 programable keys. One of the many options for programming the keys is to assign keys to be BLF keys. So I think a good solution is to label two of the programable keys VM Box 2 and VM Box 3 and set them to be BLF keys. Then when a VM is left on one of those boxes I can just have an action that turns the BLF on for the correct key. Can this be done with FS. How could I go about doing it? Thanks in advance, - Mark

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sofia deflect issue
by jonathan augenstine 06 Jan '09

06 Jan '09

I have a lua script that plays a prompt and then collects digits from an inbound call. After making a database query the call is either dropped or transferred to another server. The way I transfer the call is via the 'deflect' command. I execute that command as follows: session:execute("deflect", "user@target_server"); When I do a tcpdump, everything seems to progress as expected. I see the correct SIP messages exchanged, but after the last 200 OK is transferred Freeswitch starts sending BYE messages to the client and the client hangs up with an "Abnormal call termination". According to RFC 3515 there are no BYE messages in the protocol exchange. I have set auto hangup to false. Any ideas on what I might be doing wrong? Jonathan

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[gtalk to sip] Ringing tone
by kriko 06 Jan '09

06 Jan '09

When placing a call between gtalk and sip, one end first picks up and then ringing starts on the other side. In that time user which answers doesn't hear any tone that could indicate a call in progress and it seems a bit odd. Is it possible to play a track in the meantime, so the caller gets a feeling that something is going on? -- kriko

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using spidermonkey to get remote http status codes
by Stephen Crosby 05 Jan '09

05 Jan '09

I've been working with mod_spidermonkey and an external api that I've set up. I need to determine the status code of http responses. I've tried fetchUrl() and fetchUrlHash(), but neither seems to get that status code. Any suggestions? --Stephen

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DTMF and firewall
by Andy Ayers 05 Jan '09

05 Jan '09

Hi, I'm using freeswitch to receive incoming calls from a sip provider namely AQL. When my freeswitch box is connected directly to the internet everything works fine. When I place a firewall/router inbetween the box and the internet, the software registers with the sip provider ok and answers calls but fails to respond to in call dtmf tones. AQL advised me to make sure I was using RFC2833 which I believe I have done by setting dtmf-type in my sip profile xml to 'RFC2833'. Can anyone advise me as to what other settings I should change to make the dtmf work correctly across the firewall/router? The router is currently set to allow all traffic. Many thanks for any help you can give. regards Andy

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[dingaling] Gtalk accept friend invite
by kriko 05 Jan '09

05 Jan '09

Is it possible for dingaling to automatically accept all new friends who sent me an invite? -- kriko

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 12
by Ben Holtsclaw 05 Jan '09

05 Jan '09

- Sent from my Windows Mobile Smart Phone - ------- Original Message ------- From: freeswitch-users-request(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Sent: 1/5/09, 9:12:15 AM Subject: Freeswitch-users Digest, Vol 31, Issue 12 Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP response code in Freeswitch (Ognjen Seslija) 2. [dingaling] Gtalk accept friend invite (kriko) 3. Re: VXML support in Freeswitch (Stefan Knoblich) 4. Re: VXML support in Freeswitch (Michael Jerris) 5. Re: [dingaling] Gtalk accept friend invite (Michael Jerris) 6. Idea/Suggestion required (shehzad p) 7. Re: [dingaling] Gtalk accept friend invite (Gonzalo Servat) 8. Re: Idea/Suggestion required (Anthony Minessale) 9. Re: LDAP Integration (Vinicius Kobashi) ---------------------------------------------------------------------- Message: 1 Date: Mon, 5 Jan 2009 10:30:42 +0100 From: "Ognjen Seslija" <oseslija(a)gmail.com> Subject: Re: [Freeswitch-users] SIP response code in Freeswitch To: freeswitch-users(a)lists.freeswitch.org Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,486,503"/> Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N <saigop(a)gmail.com> wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > 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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