June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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JavaScript ApiExecute return values
by David Swardstrom 11 Jun '10

11 Jun '10

One of the problems that I have with the Wiki is that some of the APIs return a value. In fact, it is possible that many return a Success/Failure if nothing else. However, the Wiki does not specify if any data is returned. The problem sometimes is not with the specified API as listed in the JavaScript Wiki page itself but the APIs that can be invoked from ApiExecute() or with session.execute(). For example, currently I need to use the url_encode() capability. I can find it in mod_commands: <quote> url_encode Url encode a string. Usage: url_encode <string> </quote> Normally I would expect that the code would return the encoded string as a return parameter but this is not stated in the description. I.E., the Wiki entry should be: <quote> url_encode Url encode a string. Usage: encoded_string = url_encode <string> </quote> It is possible that the implementation encodes the string and returns it in the same parameter (i.e., in place). I tried looking at the code, but still am too new at looking at the FreeSwitch code. Of course, I would try it/test it and will do this. But it would be useful if the Wiki was clearer. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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Change in behaviour for ESL events?
by peely 11 Jun '10

11 Jun '10

Hi, In the latest git snapshot I've compiled, event subscriptions within ESL no longer seem to function in the same way. I used to issue "filter Unique-ID {uuid}\n\n" and "filter Other-Leg-Unique-ID {uuid}\n\n" followed by "events plain all\n\n". I did this because "myevents\n\n" would not allow me to subscribe to events for background jobs issued by bgapi, which I do quite a lot. Applying the filter then subscribing to all events seemed the most stable and allowed me to subscribe to additional events should I need them. In the latest snapshot, I don't receive any events through this mechanism. I tried "event text all" as newly suggested on the ESL outbound wiki page, but this transmits a heap of white space to my socket then kills freeswitch! Could somebody please tell me if this is something that is "work in progress" and will ultimately resume old behaviour, or should I be doing something else to monitor events for my uuid and anything I spawn in that session? Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Change-in-behaviour-for-ESL-e… Sent from the freeswitch-users mailing list archive at Nabble.com.

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filter events in Lua
by Helmut Kuper 11 Jun '10

11 Jun '10

Hello, Is there a way to filter events in Lua like this? filter Unique-ID <uuid> regards Helmut

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Git SubModule Add
by Aza Tek 11 Jun '10

11 Jun '10

Hi, Is possible to only 'checkout' specific components of FreeSWITCH as submodules using Git? If not, will this be made possible in the near future? Thanks Aza

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