- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] UNIMRCP RECOGNIZE headers
by Christopher Rienzo 07 Jan '10

07 Jan '10

Neil, In general, you can set most MRCP params in mod_unimrcp like this: detect_speech unimrcp {speech-complete-timeout=5000,speech-incomplete-timeout=5000}grammar grammar-name These params will remain set on the speech handle until freeswitch closes it. For ASR, that's when the call ends. > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Neil Burgess > Sent: Thursday, January 07, 2010 10:53 AM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] UNIMRCP RECOGNIZE headers > > Hello, > > Wondering if anyone can help with a unimrcp question. > > We are using the UNIMRCP client in FreeSwitch to communicate with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech Timeouts that are occurring on the server side, however we have been advised that the only way these can be affected is via headers in the MRCP requests. So, we need to be able to set headers such as Speech-Complete-Timeout, Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request. Is there a mechanism in FreeSwitch which we can use to pass these down before, (or as) we issue the “detect_speech” command. We are happy to use a jscript interface, or whatever if such a capability is available. > > Many thanks, > Neil > _______________________________________________ > 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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Call through gateway without register > sends to gateway name?
by peely 07 Jan '10

07 Jan '10

Hi, I have a problem where I'm trying to send calls to a gateway that does not support registration. In my external sip profile directory I have a file containing: <include> <gateway name="mygateway"> </gateway> </include> Then in my dialplan I have a dialplan with an action of: <action application="bridge" data="sofia/gateway/mygateway/1234567890"/> However, when I call out, the Sofia diag shows: sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR mygateway (to [172.16.1.1]:53) Could somebody please tell me how I get the gateway config to send INVITEs to a specific IP? Thanks, Neil. -- View this message in context: http://old.nabble.com/Call-through-gateway-without-register-%3E-sends-to-ga… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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freeswitch and H323
by Pete Kay 07 Jan '10

07 Jan '10

Hi, has anyone been able to get H323 to work? I have problem trying to get it compiled with either 1.0.4 or 1.0.5. Thanks, pete

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a weired problem when calling pstn number(s) with openzap and libpri
by Tzury Bar Yochay 07 Jan '10

07 Jan '10

Hi all, I got a weired problem and I suspect it has to do with either openzap or libpri. My setup is simple FS box, and a Sangoma E1 (A101) card. Everything seems to be well configured and working. Yet, sometimes, when I call from the SIP the call failed, on the client I get "call ended" message. On the server when looking at the log I see UNALLOCATED_NUMBER (line:274) I can certainly tell that this number exists and alive and it is not a wrong number. to make it even more complicated, calling that problematic number over and over, yields a statistics of working rate 5/1 that is every 5th or 4th call works I placed here the logs of the working case and the not working case http://gist.github.com/raw/271113/b8f13ef292669a1ab69878bfe205fc41d3722d0a/…<http://gist.github.com/raw/271113/b8f13ef292669a1ab69878bfe205fc41d3722d0a/…> http://gist.github.com/raw/271113/8b8b55922ebd8faffe1a850681f6a642dbdf4f04/… please help, Tzury

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Re: [Freeswitch-users] Zap channel not released when voicemail starts
by François Legal 07 Jan '10

07 Jan '10

Thanks. This works perfectly. François On Tue, 5 Jan 2010 10:11:05 -0600, Anthony Minessale wrote: one way is to run tone_detect on the busy signal and map it to the hangup app On Tue, Jan 5, 2010 at 3:44 AM, Ahmed Naji wrote: I'll second that. My way of dealing with it has been to write a little script to detect hangups on the TDM end, then force release the corresponding "B-leg" that is hooked up to VM. In the process of converting this to an FS module. Not clean .. but works. Would have liked to see the same code within FS core and, if appropriate, the VM subsystem to achieve the same end. Regards, Ahmed. 2010/1/5 Anthony Minessale hangup detection on TDM is a bitch. On Mon, Jan 4, 2010 at 10:07 AM, François Legal wrote: Hello, I have an issue with voicemail and openzap channels. When an incoming call on an openzap channel is bridged to voicemail, if that channel is hung up before the beginning of voicemail recording, that channel is kept open open until 3 or 4 seconds after the voicemail started to record the message. What should I do to make freeswitch/voicemail release the channel immediately when the caller hang up ? Thanks in advance François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [4] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [5] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [6] http://www.freeswitch.org [7] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [8] ClueCon http://www.cluecon.com/ [9] Twitter: http://twitter.com/FreeSWITCH_wire [10] AIM: anthm MSN:anthony_minessale@hotmail.com [11] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [12] IRC: irc.freenode.net [13] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [14] iax:guest@conference.freeswitch.org/888 [15] googletalk:conf+888@conference.freeswitch.org [16] pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [17] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [19] http://www.freeswitch.org [20] -- Ahmed Naji _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [21] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [22] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [23] http://www.freeswitch.org [24] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [25] ClueCon http://www.cluecon.com/ [26] Twitter: http://twitter.com/FreeSWITCH_wire [27] AIM: anthm MSN:anthony_minessale@hotmail.com [28] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [29] IRC: irc.freenode.net [30] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [31] iax:guest@conference.freeswitch.org/888 [32] googletalk:conf+888@conference.freeswitch.org [33] pstn:+19193869900 Links: ------ [1] mailto:a.alalousi@gmail.com [2] mailto:anthony.minessale@gmail.com [3] mailto:devel@thom.fr.eu.org [4] mailto:FreeSWITCH-users@lists.freeswitch.org [5] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6] http://lists.freeswitch.org/mailman/options/freeswitch-users [7] http://www.freeswitch.org [8] http://www.freeswitch.org/ [9] http://www.cluecon.com/ [10] http://twitter.com/FreeSWITCH_wire [11] mailto:MSN%3Aanthony_minessale@hotmail.com [12] mailto:PAYPAL%3Aanthony.minessale@gmail.com [13] http://irc.freenode.net [14] mailto:sip%3A888@conference.freeswitch.org [15] http://iax:guest@conference.freeswitch.org/888 [16] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org [17] mailto:FreeSWITCH-users@lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] mailto:FreeSWITCH-users@lists.freeswitch.org [22] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [23] http://lists.freeswitch.org/mailman/options/freeswitch-users [24] http://www.freeswitch.org [25] http://www.freeswitch.org/ [26] http://www.cluecon.com/ [27] http://twitter.com/FreeSWITCH_wire [28] mailto:MSN%3Aanthony_minessale@hotmail.com [29] mailto:PAYPAL%3Aanthony.minessale@gmail.com [30] http://irc.freenode.net [31] mailto:sip%3A888@conference.freeswitch.org [32] http://iax:guest@conference.freeswitch.org/888 [33] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org

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video
by Ken Gillett 07 Jan '10

07 Jan '10

Sorry if this is a really basic question, but how do SIP video phones work and what capabilities does FreeSwitch have in this regard? Does a PBX such as FS have to deal with the video stream, or does it just pass around a URL that the phone then uses as the source for the streaming video? Or is the video data all 'switched' along with the audio data? How would conference calls be handled? Is this part of the SIP standard, or is it all proprietary and just down to the phone designer/manufacturer? Ken G i l l e t t _/_/_/_/_/_/_/_/

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encrypt voicemail emails and attachments?
by Harondel J. Sibble 07 Jan '10

07 Jan '10

Looking throught the wiki, I see various configs for having FS email you a copy of received voicemail messages, has anyone done any work with having the voicemail messages gpg encrypted with public prior to sending? Or is that something that should pretty much be handled at the mta level leaving FS out of the mix altogether? I'm thinking probably so, but before I try to do this, I figured i'd ask first. -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help(a)pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice)

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Script for Presence using PHP
by Edmar Cruz 07 Jan '10

07 Jan '10

Dear Sir, Is there a script to trigger whether a user is online or offline using a script or a PHP code? Presence for freeswitch is for command line only? Can you give me an example? Thanks, Edmar -- View this message in context: http://old.nabble.com/Script-for-Presence-using-PHP-tp27050100p27050100.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Codec Negotiation: Codec telephone-event:101:8000:20
by Mark Campbell-Smith 07 Jan '10

07 Jan '10

Hi! If I try to call out on one of my voip providers I get INCOMPATIBLE_DESTINATION. Something is going wrong with codec negotiation: 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3262 Set 2833 dtmf payload to 101 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [NOTICE] sofia.c:3937 Hangup sofia/external/020555500 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] According to their website, the codec they us is G711 a-law or G729. What codecs have name CN and telephone-event? How do I get these to match? I assume the format of the debug output is: Audio Codec compare [Codec Name:Media Format:Media Name:Rate:RTP Size?] Thanks

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Is there support for custom fields for all events
by Jon Bruel 06 Jan '10

06 Jan '10

I would like to add a custom field to all events sent from the FS. The value of the field should be the value of a global variable, the name should preferably be the name of this variable. Is there any way to set this up? /Jon

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