June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch / Webserver
by Rudolf Denert 18 Jun '09

18 Jun '09

Yes, I removed the tags but with no effect. I think the problem is that the webserver doesn't look in the directory where featuers.xml is deposited on the freeswitchserver (/opt/freeswitch/conf/dialplan/). The issue is that FS finds the context when dialplan is the directory /opt/freeswitch/conf/dialplan/public/ . But when it is on the webserver (it is on another server with a different IP-address) I get the error that I told you. What should i verify in the default config. Greetz ----- Ursprüngliche Mail ----- Von: "Brian West" <brian(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Mittwoch, 17. Juni 2009 15:47:57 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Freeswitch / Webserver Its clearly telling you that context features doesn't exist... did you remove the context tags around your extension so that it would be in the correct context? Review the default config again. /b On Jun 17, 2009, at 8:42 AM, Rudolf Denert wrote: > Context features not found _______________________________________________ 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 -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert(a)tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enthält vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

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Fwd: [UniMRCP] Open source ASR and TTS plugins
by Brian West 18 Jun '09

18 Jun '09

Guys I'm tossing in $250 dollars of my own money on this ... who is going to pitch in? Arsen and I have been talking about how to accomplish this so we'll have an Open Source Speech Server via MRCP. Who wants to pitch in please paypal brian(a)freeswitch.org and I can wire it to Arsen. Thanks, Brian Begin forwarded message: > ----- Forwarded Message ---- > From: Arsen Chaloyan <achaloyan(a)gmail.com> > To: UniMRCP <unimrcp(a)googlegroups.com> > Sent: Wednesday, June 17, 2009 10:57:30 PM > Subject: [UniMRCP] Open source ASR and TTS plugins > > Anybody interested in the development of open source ASR and TTS > plugins for UniMRCP server write me offlist. > > PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ > Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ > > Thanks, > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org

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VAD, TALK and NOTALK events
by Steven Brown 18 Jun '09

18 Jun '09

Hi, I have been trying to pick up TALK and NOTALK events but with no success, I have enabled VAD for "both" in my config and the rtp is stopping and starting as expected however when I hook up to the event socket and request "event talk notalk" nothing is ever fired, any thoughts on where I am going wrong appreciated. Thanks Steve Steven Brown email steve(a)justfone.com office 08707706968 mobile 07768755409 fax 07884636663 Justfone - Company Reg. No. : 3926817 Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW The contents of this e-mail may be privileged and are confidential. It may not be disclosed to or used by anyone other than the addressee(s), nor copied in any way. If received in error, please advise sender, then delete it from your system. Internet email communications are not secure and therefore Justfone do not accept legal responsibility for the contents of this message. Any views or opinions presented are solely those of the author and do not necessarily represent those of Justfone unless otherwise specifically stated.

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Error on mod_nibblebill cannot connect to ODBC
by Edmar Cruz 18 Jun '09

18 Jun '09

Hi I experiencing an error on mod_nibblebill. I already load it from autoload_configs, especially mod_spidermonkey. Uncomment mod_spidermonkey_odbc. I also download unixodbc and created the files /etc/odbcinst.ini and /etc/odbc.ini with the correct format [zenoss] DATABASE = tcapi USER = root PASS = password ..... I type also on the console isql zenoss root password. Also working... But an error occur on freeswitch Cannot connect to user [root] ... What do you thinks is the problem? -- View this message in context: http://www.nabble.com/Error-on-mod_nibblebill-cannot-connect-to-ODBC-tp2404… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Weird (or expected) behaviour on record/playback using ESL?
by Peter Olsson 18 Jun '09

18 Jun '09

I'm not quite sure if this is the expected behaviour, I just wanted to make sure. I've developed a simple IVR application using event socket. I dial in to the dialplan and park the call, and then I let the IVR application do whatever it's supposed to. I basically listen for DTMF events and play and record files. Today I just noticed that if I issue a "api uuid_record <uuid> start <filename>", and then do a file playback (using SendMsg, with call-command execute and execute-app-name playback), the playback is sent both to the caller, and to the recorded file. Is this the way it's supposed to work, or should I playback files in another way? /Peter

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SIP and nat traversal IPv4/IPv6 issue
by Jason White 18 Jun '09

18 Jun '09

this is an issue which I've been discussing with Brian West on IRC and in e-mail correspondence, which I thought I should bring to the list so that others can look at it as well. The configuration My external SIP profile has its ext-sip-ip and ext-rtp-ip set to stun:stun.freeswitch.org. This is necessary for nat traversal. I have an internal-ipv6 profile as well, which is working, but for some reason it's interfering with calls on the external profile (which of course is an IPv4 profile). The symptom is the following line in outgoing SIP messages while attempting to establish a call to a gateway via the external profile: o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org in response to which, the other side returns a 488 (invalid session description). I can confirm that ext-sip-ip and ext-rtp-ip are *not* set in the internal-ipv6.xml profile, but they are said as explained above in the external profile, where the problem lies. This looks like a bug to me but I want to rule out misconfiguration first. This has been tested on revision 13806.

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Re: [Freeswitch-users] Controlling Conference Controls
by Jason Cromes 18 Jun '09

18 Jun '09

I haven't gone to the trouble (yet) of making this work, but I believe you could use execute_application from the conference controls to do just about anything with JavaScript... This Wiki page has some good JavaScript examples... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR There are a couple functions in there for voicing user count, etc. So, I *believe* you could stick those in a script by themselves and call them using execute_application out of the caller controls... Somehow, you would have to identify what user is calling the script and what conference they're in. (Once you're in the javascript, you could check a "conference number" variable that you set when the person entered the conference. Or, you could parse the output of "conference list" until you found that session's UUID.) I don't know what else you're trying to do, but once you get one of them working, the rest should follow a similar template. Post back if you make it work, I'm interested!

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Segmentation fault with record_session
by Jingwei Yang 18 Jun '09

18 Jun '09

Hi Team, I've been using the record_session feature to record call sessions. Here's how I prepared the dialplan: <extension name="skypiax"> <condition field="destination_number" expression="^2909/(.*)$"> <action application="record_session" data="/tmp/data.wav"/> <action application="bridge" data="skypiax/ANY/$1"/> </condition> </extension> And here's how I trigger it: *freeswitch(a)localhost.localdomain>originate skypiax/skypiax2/userAAA 2909/userBBB* The call can be established and the data.wav file was generated without any problem. However, once userAAA hung up, a segmentation fault occurred and freeswitch was automatically shut down. Here are what I got from the console: *freeswitch(a)localhost.localdomain> originate skypiax/skypiax2/userAAA 2909/userBBB 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b] 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/skypiax2/userAAA 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/skypiax2/userAAA] has been answered 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer() Transfer skypiax/skypiax2/userAAA to XML[2909/userBBB@default] API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output: +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b freeswitch(a)localhost.localdomain> 2009-06-15 17:25:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH->2909/userBBB in context default 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a] 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing() Ring-Ready skypiax/ANY/userBBB! 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered() Channel [skypiax/ANY/userBBB] has been answered 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA [CS_DESTROY] 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY] Segmentation fault (core dumped) * Please kindly let me know whether there's anything wrong with the dialplan or the way how I originated the call. Thanks! -Jingwei **

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 172
by Steven Brown 17 Jun '09

17 Jun '09

You can also send and recieve SMS on the PORTech Gateways via AT commands over a socket connection, I've not checked but would assume all other AT commands will work so you can check signal strength etc this way also. Steve Message: 4 Date: Wed, 17 Jun 2009 23:31:06 +0200 From: Jan Kubr <jan.kubr(a)gmail.com> Subject: Re: [Freeswitch-users] Which GSM gateway to buy? To: freeswitch-users(a)lists.freeswitch.org Message-ID: <698401620906171431t10432015xa78976a401dc5c5b(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We plan to buy one of these: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterpris… since you can use SMTP/POP3 to manage SMS. Jan

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high latency
by David Burgess 17 Jun '09

17 Jun '09

Greetings, I'm new to freeswitch, just playing with it in a home/small office at present. Overall I've been really impressed with it. One issue we've noticed is a pronounced latency on some or perhaps all calls over roughly 500 seconds in duration. The latency is approximately 3 seconds or more round trip. I tried setting jitterbuffer_msec=20 but it has not prevented the problem. The call quality is otherwise good with no noticeable choppiness or jitter. Other local network traffic appears to be irrelevant, as in the latency seems to occur even when the LAN is idle or experiencing only sporadic bursts of activity, like typical browsing. I read the FAQ and searched jitter, lag and latency in the wiki and list archive but didn't come up with anything. Is this a known issue? Could it be that the problem is specific to my platform? I'm running the freeswitch package on pfsense (FreeBSD-based) and nobody in the pfsense forums seems to know the cause or solution, although others have reported the same issue. Much thanks for your input, db

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