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uuid_media hangs
by peely 14 Jun '10

14 Jun '10

Hi, I'm trying to use the uuid_media command to control the media path after a call has been established. I can't seem to get it to respond wither through the console or through ESL. Whenever I issue a uuid_media {uuid} or uuid_media off {uuid} the console or connection just hangs, even when I have debug output I get nothing. I've tried this with both bridged and originated calls. I have enable-3pcc set to true on the profile. Effectively I am trying to set up an IVR through ESL then connect the caller on to a destination, once connected I want to push the media away. Looking at the SIP coming in from the far end I see invite and update is allowed so in theory this should be supported by the remote party. Could somebody please let me know what the conditions are to be able to use uuid_media? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p51… Sent from the freeswitch-users mailing list archive at Nabble.com.

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updated sqlite3?
by Helmut Kuper 14 Jun '10

14 Jun '10

Hello, today I played around with lua scripts using lua-sqlite3 module. I found that the delete statement of the sqlite3 version (3.1.3?) FS is using is not able to handle LIMIT in DELETE statements. Also the centos sqlite3 cli tool (v3.3.6) doesn't do. I found that sqlite3 version 3.6.23 does the trick as long as you use SQLITE_ENABLE_UPDATE_DELETE_LIMIT during compile time. So any chance to get FS sqlite sources updated? As long as not, can I riskless update FS's sqlite source to 3.6.23 and recompile FS? regards Helmut

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Non-Blocking Music on Hold
by Jon Davies 14 Jun '10

14 Jun '10

Hi There I'm attempting to port an application I've written in CCXML/VXML to FreeSwitch/javascript and finding it a very enjoyable experience. I've hit the limits of the documentation and my knowledge and was wondering if there were any clever people on the list who can assist me with one little part. My scenario is: Make an outbound call and play it MOH Make a 2nd outbound call If the 2nd call connects, take the 1st call off MOH and bridge them together If the 2nd call fails for whatever reason, take the 1st call off MOH and play it a 'sorry' message. My problem is, all my attempts to play MOH to the first call results in blocking, meaning I dont ever get to originate my 2nd call. I've tried firstcall.streamFile("music.wav"); and firstcall.execute( "fifo", "myqueue in undef 'music.wav'" ); but both block until I hang up. Anyone got any insight as to how I can achieve this scenario in a non-blocking manner? I'm using javascript. -- Regards Jon Davies

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Full NAT bypass solution with STUN/ICE/FS but without TURN: is possible?
by Fabio Pietrosanti (naif) 14 Jun '10

14 Jun '10

Hi all, i need to setup a server that handle VoIP connection in a way that always works in the most efficient way respect to latency and NAT traversal. FS provide very good NAT detection system but unfortunately when it fail, it just fail and does not have any kind of fallback mechanism. If FS do bypass-media and the client cannot communicate because the phone call is lost. If FS do proxy-media, the clients can always communicate (over UDP) but there is added latency. So we cannot rely only on FS NAT detection system and need also some client side system. Now, in theory the best approach would be not to touch FS but just use client-side system based on ICE/STUN with a TURN server-side component as a fallback. However there's no widely available TURN server, or at least TURN is not widely and well diffused just now. I am wondering whether it may be possible to make a setup where: - Client try to establish peer to peer connectivity with STUN/ICE (so FS is doing bypass-media not touching the SDP) - If they cannot work, will go in fallback to FS (but with proxy-media) instead than going via TURN So using the FS B2BUA proxy-media feature as alternative to TURN server, but letting clients to dynamically try to connect directly one each other by using the ICE methods that are dynamic. Does it seems something feasible? Fabio Pietrosanti

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leg_timeout isn't working.
by Nagalenoj H. 14 Jun '10

14 Jun '10

Dear friends, I've posted an issue in jira(2 days back) but I didn't get any response there. I just want to confirm whether any one else is facing the same and it is really an issue. Description: When I tried to execute 'bridge [leg_timeout=10]user/1010', it doesn't quit ringing if the callee didn't respond in 10 seconds. But when I use it in {leg_timeout=10}user/1010, it's working. When I refer the wiki, it is given as it shouldn't be used in curly braces. 'Can be used in per-leg [], but not in global {} for the dialstring.' -- From wiki (http://wiki.freeswitch.org/wiki/Channel_Variables#leg_timeout) Attached the log here, http://jira.freeswitch.org/browse/MODAPP-433 -- Regards, Nagalenoj H.

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Missing CDRs with ATTENDED TRANSFER
by katarina djakovic 13 Jun '10

13 Jun '10

Dear Freeswitch users, we have a problem where no CDRs are created for assisted transfer. We are running FS 1.0.6. We have extensions and one gateway to connect to PSTN. The scenario is the following: 1) Extension A calls a PSTN number (through the gateway) lets say nubmer C. Extension A is using Twinkle soft phone. Connection gets established. 2) Then, using Twinkle, extension A chooses to transfer the call to extension B by pressing xfer button in Twinkle's GUI. A calls B. 3) Then A hangs up (presses xfer button again) while connection gets established between B and C. However, FS produces only 2 CDRs for the whole process. Namely, there is a CDR for the step 1) A -> C. There is a CDR for 2) A -> B, but there is no CDR for 3) B -> C. The two CDRs are produced just after the call is transfered, but the third CDR never gets produced. Although, the FS is aware of the ATTENDED_TRANSFER as this appears as the 'reason' in the second CDR. Here are the CDRs produced: "2000","2000","gateway","external",""Katarina Piljanovic" <2000>","sofia/external/2000@95.180.91.118:5080","sofia/external/381117543472","bridge","sofia/g ateway/provider/381117543472","2010-06-10 16:52:11","2010-06-10 16:52:17","2010-06-10 16:55:58","227","221","NORMAL_CLEARING","","eaf26c9d-4cd7-4f65-a53e- a426b50da52d","","","GSM","GSM","none","none" "2000","2000","2002","external",""Katarina Piljanovic" <2000>","sofia/external/2000@95.180.91.118:5080","sofia/external/2002","bridge","{disable_rtp_au to_adjust=true}sofia/external/2002%95.180.91.118","2010-06-10 16:53:31","2010-06-10 16:53:33","2010-06-10 16:56:00","149","147","ATTENDED_TRANSFER","","ae6547cb-983d-405b-a4c1- 1afcd3260c4d","","","GSM","GSM","none","none" Is there a way to get the third CDR, is it something in config that we should set? As, this way the B and C can talk forever and never get billed for the call. Do you need to see the Dialplan to suggest the resolution? Also, the first CDR is not correct either, as it should list the number called instead of the word "gateway" in the third coloumn, as it normally does when we call through the gateway. Many thanks in advance, Katarina _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

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Playback of Multiple Items
by broken dash 13 Jun '10

13 Jun '10

Hello, Does there happen to be a way to configure FS to playback multiple shoutcast streams to an endpoint? I've tried making two extensions setup like so... My endpoints can dial each single ext without issue but the combine ext only seems to grab the first bridge xml entry and I've tried several variations of bridging multiple endpoints, etc.. Perhaps I'm approaching this incorrectly? Thanks, Brian M. == <extension name="273" > <condition field="destination_number" expression="^273$"> <action application="answer"/> <action application="playback" data="shout://relay.someserver.com/feed01"/> </condition> </extension> <extension name="3973" > <condition field="destination_number" expression="^3973$"> <action application="answer"/> <action application="playback" data="shout://relay.someserver.com/feed02"/> </extension> <extension name="combine" > <condition field="destination_number" expression="^combine$"> <action application="bridge" data="sofia/internal/273@${domain}"/> <action application="bridge" data="sofia/internal/3973@${domain}"/> </condition>

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DETECTED_SPEECH events while playing a file
by Jan Kubr 12 Jun '10

12 Jun '10

Hi, I'm trying to detect speech while playing a file over the event socket. I start detecting speech: call-command: execute execute-app-name: detect_speech execute-app-arg: pocketsphinx ivr_grammar ivr event-lock:true Then playback the file: call-command: execute execute-app-name: playback execute-app-arg: ivr.wav event-lock:false Then resume the speech detection: call-command: execute execute-app-name: detect_speech execute-app-arg: resume event-lock:true Now when I say something, I do see this in the FreeSWITCH console: 2010-06-10 23:29:06.994346 [DEBUG] mod_pocketsphinx.c:383 Recognized: ONE, Confidence: 100 But I don't receive the DETECTED_SPEECH events over the socket. However, if I wait until the file is played, that is until I see this in the FreeSWITCH console: 2010-06-10 20:48:50.512148 [DEBUG] switch_ivr_play_say.c:1428 done playing file and say something after that, I do get the DETECTED_SPEECH events. Is there a way I could receive these events while the file is being played? Thanks, Jan Kubr

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calling between extension
by budi wibowo 12 Jun '10

12 Jun '10

hi i have question regarding profile user in FS. in /usr/local/freeswitch/conf/directory/sip1.domain.com i have budi.xml containing <include> <user id="budi"> <params> <param name="password" value="1000"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> and i have other profile for other user say james.xml. what need to be configured from FS so i can can call james directly from my softphone regards

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Andres Martin has invited you to Dropbox
by Dropbox 11 Jun '10

11 Jun '10

We're excited to let you know that Andres Martin has invited you to Dropbox! Andres Martin has been using Dropbox to sync and share files online and across computers, and thought you might want it too. Visit http://www.dropbox.com/link/20.9kABIoVFR8/NjIxODA4MDYyNw to get started. - The Dropbox Team ____________________________________________________ To stop receiving invites from Dropbox, please go to http://www.dropbox.com/bl/070b00e90e9d/freeswitch-users%40lists.freeswitch.…

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