January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video?
by Greg Millam 20 Jan '12

20 Jan '12

Hi - The video refresh request does the job. recorded videos are very good quality, now, and start only when record_fsv is called. Thank you very much! - Greg Millam > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 18 Jan 2012 13:41:49 -0600 > Subject: Re: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? > try this: > > commit 67f559685f97b273dc5f5d23b69cc85cf90e32b0 > Author: Anthony Minessale <anthm(a)freeswitch.org> > Date: Wed Jan 18 14:08:55 2012 -0600 > > req vid refresh in fsv > > > On Wed, Jan 18, 2012 at 11:33 AM, Greg Millam <greg(a)brilliantecho.com> wrote: >> >> > commit c358f67fe4348b8b5209328660aef02d8ffaf15f >> > Author: Anthony Minessale <anthm(a)freeswitch.org> >> > Date: Tue Jan 17 12:19:23 2012 -0600 >> > >> > eat inbound vid while playing fsv files >> > >> > >> > I didn't have a way to test it setup so i'll need feedback. >> >> It works, but now since it's missing the first frame, the resulting >> video is only partial, showing new bits. Is there a way to send a >> message to the client to "clear and send a full frame of video" ? >> >> > On Mon, Jan 16, 2012 at 11:38 PM, Greg Millam <greg(a)brilliantecho.com> wrote: >> >> >> >> Hi folks - >> >> >> >> I have a freeswitch dialplan + script that first calls play_fsv to >> >> play a greeting, then record_fsv to record incoming video to fsv. >> >> >> >> It works fine, but there's one issue: Apparently, freeswitch is >> >> buffering incoming video during play_fsv. When record_fsv is called, >> >> that buffer is dumped into the .fsv file, resulting in several seconds >> >> of unneeded video. >> >> >> >> Is there a way to empty that incoming video buffer before record_fsv >> >> begins recording? >> >> >> >> Thank you! >> >> >> >> - Greg Millam

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Modify sip_invite_req_uri, sip_invite_to_uri
by Brian Foster 20 Jan '12

20 Jan '12

In this example: send 1149 bytes to udp/[<extension_ip>]:5060 at 14:47:46.972424: ------------------------------------------------------------------------ INVITE sip:<extension>@<extension_ip> SIP/2.0 Via: SIP/2.0/UDP 204.93.201.164;rport;branch=z9hG4bKDZKU0eQBtr3Be Max-Forwards: 64 From: "UNKNOWN" <sip:<caller_id_number>@<FS_ip>>;tag=FFr3B9gSB0KZK To: <sip:<extension>@<extension_ip>:5060> ... I need to change sip_invite_req_uri and sip_invite_to_uri so that the switch downstream can route the call based on the DID they have with me. I've tried setting this several different ways but no avail. I'd like it to be global if possible. Is there a good spot to set these variables? -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

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Log-rotation and uploading of .csv files.
by Henrik Aagaard Sørensen 20 Jan '12

20 Jan '12

I'm in the beginning of development for billing on my FS instance. Currently I'm just in the log-rotation and uploading of the .csv-files stage. I'm having trouble understanding completely how this should work, to be "bullet-proof" in a production environment. I've read the "Example Perl Script for CDR into MySQL" in http://wiki.freeswitch.org/wiki/Mod_cdr_csv. What I would like to accomplish is uploading the Master.csv-file to another server, which should take care of all the billing, so it doesn't interfere with the VoIP-server. But how can I make sure, that when I'm uploading the Master.csv file, that FS isn't currently writing to the file as well. I'm thinking of moving the Master.csv to Master-YYYY-MM-DD-HH-MM.csv every minute and then uploading those files to another server. But then again, how can I make sure, that when moving the file to Master-YYYY-MM-DD-HH-MM.csv that FS isn't writing to the file? Can someone share some light on this matter?

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Running a BASH script within a dialplan
by Brian Foster 20 Jan '12

20 Jan '12

Hola, I'm trying to run a BASH script which turns sends a message to Google's Transcription API. It's a modified version of the example posted on the wiki. Here we go: !. #!/bin/sh 2. sox $1 message.flac pad .1 0 rate 16k 3. wget -q -U "Mozilla/5.0" --post-file message.flac --header="Content-Type: audio/x-flac; rate=16000" -O - " http://www.google.com/speech-api/v1/recognize?lang=en-us&client=chromium" > message.ret 4. echo "$(cat message.ret | sed 's/.*utterance":"//' | sed 's/","confidence.*//')" 5. rm message.flac 6. rm message.ret 7. # rm $1 Before I stick that in a dialplan, I've been trying to run it within fs_cli to make sure it works. However, I get: -ERR no reply ...every time I run it. Is there some fundamental flaw that FreeSWITCH doesn't recognize as an output? I plan on putting together a few BASH scripts and a dialplan to basically do the same thing that Mundy has done with "Siriously". -BDF -- This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

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Urgent Need for a FreeSWITCH Developer in Sunny San Diego! (This is contract to hire opportunity and the client is looking to fill this role ASAP.)
by Stewart, Justin 19 Jan '12

19 Jan '12

HAPPY NEW YEAR! Thank you for taking the time to read this email... Our client is in need of a Sr. Level Software Engineer, who has strong embedded or networking software experience, and has built Windows code in C (an ideal candidate will be a developer who has worked on an Asterisk project, or has strong Asterisk experience). They will be producing simple GUI apps and simple install packages, so a windows GUI expert would not be the right person. If this person has telecom experience (Lucent, Nortel, Avaya, Cisco, other telecom etc and knows any one of the protocols SIP/MGCP/H323/etc) who has also built Windows apps is the best bet. Linux experience is VERY beneficial. Software Engineering skills C programming in Windows Visual Studio 2008 IDE Windows application packaging/installers (MSI) Network programming in Windows using winsock Application specific skills Needed Configuring FreeSWITCH or Asterisk on Windows XP/Server 2008 Installing FreeTDM drivers (for Windows) Implementing TCP/UDP IPv4 network applications in C Knowledge of VOIP Signalling protocols (either SIP, SCCP, MGCP, or H.323) Testing TCP/UDP network applications using packet impairment & network simulation This is contract to hire opportunity is located in Carlsbad, CA and the client is looking to fill this role ASAP. Compensation: Negotiable based on relevant experience. Please contact me at jstewart(a)teksystems.com<mailto:jstewart@teksystems.com> or 858 320 2767 Regards, Justin Stewart, Account Manager 4180 La Jolla Village Drive, Suite #100 La Jolla, CA 92037 Toll Free: (888) 260-6282 Cell: (619) 318 8564 Office: (858) 320-2767 Fax: (858) 320-2790 http://www.teksystems.com/Locations/United-States/California/San-Diego.aspx [cid:image001.gif@01CCD680.949C4910]<http://www.teksystems.com/> ________________________________ This electronic mail (including any attachments) may contain information that is privileged, confidential, and/or otherwise protected from disclosure to anyone other than its intended recipient(s). Any dissemination or use of this electronic mail or its contents (including any attachments) by persons other than the intended recipient(s) is strictly prohibited. If you have received this message in error, please notify us immediately by reply e-mail so that we may correct our internal records. Please then delete the original message (including any attachments) in its entirety. Thank you.

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Re: [Freeswitch-users] anonymous calls
by Rodney 19 Jan '12

19 Jan '12

I am using fusinpbx on windows which logs everything I suppose. I open the csv and it has multiple columns. one is caller id name and one is caller id number. both show "anonymous". some calls work some don't. this also appears when monitoring a conference with the fusion web panel and when i watch the console when the call comes in it also says mod_dialplan_xml.c:481 Processing Anonymous <Anonymous>->DID (masked for troubleshooting) in context public the provider log shows 2012-01-19 16:34:43-06 FULL ANI (masked for troubleshooting) DID (masked for troubleshooting) 00:08 ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, January 19, 2012 4:42 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 183 ----- Forwarded Message ----- 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: Play media dialplan (Michael Collins) 2. announce count to user only entering conference - Parse Error (Rodney) 3. anoymous calls (Rodney) 4. Re: anoymous calls (Brian West) Each extension can record and select his/her own greeting from the advanced options menu. -MC On Thu, Jan 19, 2012 at 9:08 AM, <dan(a)subformat.net> wrote: Thanks MC, will do. > >So I take it this sort of functionality should work then? Im not trying to >do something out of the ordinary? > >Is there anyway to specify a different voicemail greeting per extension? >Might be able to get round it that way. > >Thanks > >Dan. > >> This is a strange one. I'd update to latest git and re-test. If you can >> consistently reproduce these symptoms after updating I would open a Jira >> ticket. >> >> -MC >> >> On Wed, Jan 18, 2012 at 3:35 AM, <dan(a)subformat.net> wrote: >> >>> The sound file does work, as i can just set the dial plan to go straight >>> to play the media and it plays fine. Its only when the play media is in >>> the middle of the dial plan that it doesnt seem to work. >>> >>> Should this work? Or should i be achieving this from a different >>> setup/dialplan? For instance, maybe specifying the voicemail file (wav) >>> to >>> play for a certain extension number? >>> >>> Thanks >>> >>> Dan. >>> >>> > On Tue, Jan 17, 2012 at 2:03 PM, <dan(a)subformat.net> wrote: >>> > >>> >> Aaah sorry! That shows just how new I am to FS. >>> >> >>> >> Heres the rather large log: http://pastebin.freeswitch.org/18151 >>> >> >>> >> Thanks again. >>> >> >>> >> Dan. >>> >> >>> >> >>> > Well, no new info there. I would confirm that the sound files >>> themselves >>> > are not broken. You might want to clean out the sounds directory and >>> then >>> > re-run "make cd-sounds-install" >>> > >>> > -MC >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting(a)freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> > http://www.cudatel.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > 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 >>> > >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means. scenario: When I enter a room with people the system says "there are" say count "other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room. this works fine when there is at least 1 other caller in the room. but when there isn't another caller and the room is empty. it says "there are" say count is a parse error so no digit read (need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero" ERR] mod_say.c:130 Parse Error! Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you. <extension name="room1"> <condition field="destination_number" expression="^301$"> <action application="set" data="conference_user_list=|"/>  <action application="set" data="conf_count=${conference 301-127.0.0.1 list count}"/> <action application="log" data="INFO ${conf_count}"/> <action application="say" data="en number pronounced ${conf_count}"/> <action application="playback" data="C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/othercallers.wav"/>  <action application="conference" data="301-127.0.0.1@default"/> <action application="transfer" data="300 xml default" /> </condition></extension> I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix? What exactly are you logging? On Jan 19, 2012, at 3:23 PM, Rodney wrote: I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org http://www.freeswitch.org _______________________________________________ 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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anoymous calls
by Rodney 19 Jan '12

19 Jan '12

I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix?

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Play media dialplan
by dan＠subformat.net 19 Jan '12

19 Jan '12

Hi guys, I'm new to Freeswitch and am currently looking to setup a dialplan that basically does the following: Ring an extension for 10 secs if not answered, play media file. After that has played, go to voice mail. This doesn't seem to work though, it goes straight from ringing the extension to voicemail and misses out the media file. If you setup the dialplan to just play the media file it plays fine, just not when its in the middle of a dialplan. <include> <extension name="public_did"> <condition field="destination_number" expression="number"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/internal/1000@<IP_ADDRESS>"/> <action application="answer"/> <action application="set" data="continue_on_fail=true"/> <action application="playback" data="/usr/local/freeswitch/sounds/<FILE_NAME>.wav"/> <action application="voicemail" data="default <IP_ADDRESS> 1000"/> </condition> </extension> </include> If you could point me in the right direction, that would be great. Thanks

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Freetdm + OpenR2
by Carlos Cesario 19 Jan '12

19 Jan '12

Hi guys! I'm try configure freeswith with E1 link R2 protocol, but I'm getting a lot errors when freeswitch services start. I have not found nothing about theses errors. Somebody have any idea about fix this ? All configsare made based inthe freeswitch wiki Errors Logs -> http://pastebin.freeswitch.org/18167 Freetdm.conf -> http://pastebin.freeswitch.org/18168 Freetdm.conf.xml -> http://pastebin.freeswitch.org/18169 /etc/dahdi/system.conf and dahi_status -> http://pastebin.freeswitch.org/18170 greats, Carlos

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feature list
by Bob Smith 19 Jan '12

19 Jan '12

I do have a wiki login actually... ;-) The reason for the post was that I'm not intimately familiar with the code tree, and hence there's a risk I'd link to the wrong thing .... I get 12 pages of results when I Search the git repository for the word "changelog". Earlier Anthony said "the change log in the build root" ... but the one at http://fisheye.freeswitch.org/browse/freeswitch.git/ChangeLog hasn't been updated since 2007 ? So I'm guessing http://fisheye.freeswitch.org/browse/freeswitch.git/docs/ChangeLog .... but that's in docs.... not the root ... ;-) >Sure go ahead and do that. You can create a login on the wiki and do that if you see that as a good addition

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