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Re: [Freeswitch-users] announce count to user only entering conference - Parse Error (Michael Collins)
by Rodney 22 Jan '12

22 Jan '12

MC, thank you for adding this macro. it seems to work fine. for my current purposes i just created silent wavs until i need the complete voice instructions. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, January 21, 2012 9:32 AM Subject: FreeSWITCH-users Digest, Vol 67, Issue 195 ----- 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: mod_rtmp + Akamai stream push (Anthony Minessale) 2. Re: announce count to user only entering conference - Parse Error (Michael Collins) 3. Re: G.722 negotiation issue with AVM Fritzbox (Brian West) I don't recall any initiative being done to facilitate what you describe but its open source code so nothing's impossible. On Thu, Jan 19, 2012 at 4:08 PM, Jock McKechnie <jock.mckechnie(a)gmail.com> wrote: Greetings wise(r) FreeSWITCH users; > >We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP >media conversion (audio only) and have been utterly delighted by how >easy and (so far) reliable this has been to both set up and run on a >daily basis. One of my superiors has gotten all excited and wonders if >we can use the same mechanism to do a stream push to Akamai for >content delivery amongst many thousands of RTMP clients (think radio >broadcast). > >It appears that the FreeSWITCH mod_rtmp module is _not_ capable of >this - it can send a fresh RTMP stream to a registered client, but it >cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I >correct in this assertion, or am I misunderstanding the documentation >(and google hits) available? > >My thanks to all; > > - Jock > >_________________________________________________________________________ >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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 I would use a phrase macro for this. In fact, this seems like such an obvious thing to do that I added a phrase macro to conf/lang/en/ivr/sounds.xml. Update to the latest latest like two seconds ago git and you will have it available. (You'll need to copy it over from the source directory to your install directory.) The new macro is called other_callers_in_conf. In your example the extension would be this: <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 ${conf_count}"/>  <action application="playback" data="phrase:other_callers_in_conf:${conf_count}"/> <action application="conference" data="301-127.0.0.1@default"/> <action application="transfer" data="300 xml default" /> </condition> </extension> The phrase macro does all the work. NOTE: I need to record a few new phrase files and then tweak the macro. In the meantime I encourage you to look at the macro that I added and tailor it to your needs. -MC On Thu, Jan 19, 2012 at 12:45 PM, Rodney <notlikeme75(a)yahoo.com> wrote: 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> > > > > > > > > > > >_________________________________________________________________________ >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 > > What is your codec selection set to in FreeSWITCH and what rev of FreeSWITCH are you running? On Jan 20, 2012, at 4:09 PM, Roland Hänel wrote: OK you're right, I read the RfC and in fact a zero port means that the >stream is rejected. However the whole "negotiation" is still broken. If FS >rejects the incorrect G.722, OK, but what about all the other (perfectly >valid) codecs in the request (PCMA, PCMU)? > >Because FS only rejects G.722, the call is doomed to fail, even though it >could work nicely if FS just selected one of the acceptable offers. > >Roland -- 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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Privacy on shared lines
by Darcy 21 Jan '12

21 Jan '12

Is there a setting to set lines to private on shared lines in the freeswitch I am using snom phones and have the shared lines working except that I would like to automatically default to private lines versus allowing barge in. Darcy Primrose

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MySQL type of read_codec and write_codec in CDR's.
by Henrik Aagaard Sørensen 21 Jan '12

21 Jan '12

I've seen the SQL script, cdr.sql, in the src-directory which should match the CDR's. But it is missing the read_codec and write_codec. I guess it should be VARCHAR's in MySQL, but can anyone verify that and what is the longest possible string (VARCHAR(80)?)?

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announce count to user only entering conference - Parse Error
by Rodney 21 Jan '12

21 Jan '12

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>

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mod_rtmp + Akamai stream push
by Jock McKechnie 20 Jan '12

20 Jan '12

Greetings wise(r) FreeSWITCH users; We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP media conversion (audio only) and have been utterly delighted by how easy and (so far) reliable this has been to both set up and run on a daily basis. One of my superiors has gotten all excited and wonders if we can use the same mechanism to do a stream push to Akamai for content delivery amongst many thousands of RTMP clients (think radio broadcast). It appears that the FreeSWITCH mod_rtmp module is _not_ capable of this - it can send a fresh RTMP stream to a registered client, but it cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I correct in this assertion, or am I misunderstanding the documentation (and google hits) available? My thanks to all; - Jock

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RTMP from non-flash/flex code
by jahnunya＠gmail.com 20 Jan '12

20 Jan '12

Hi. I'd like to ask if anybody succeeded to connect streams from RTMP session to a call, for example SIP, not using flex/Flash. I need a way to connect a SIP call to voice streams available from RTMP server (Red5). Which side initiates the call doesn't matter to me. I tried the flex client and it works fine: a call from Flash/flex could initiate a SIP call in FreeSWITCH. However, I need to connect streams from RTMP server (Red5). So far, every attempt to do the same failed: I can connect to FreeSWITCH via RTMP, of course, and make a call (by calling the "makeCall" method) but the streams cannot be connected. I tried the other way round: from SIP to RTMP, but it seems like the module was not meant to work this way, as it complaints about the session id/uuid. Unfortunately, I am not a C/C++ developer, so reading the code doesn't help me. Any solution or explanation of the RTMP module will be highly appreciated. Thanks in advance

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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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