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stop/start RTP
by Tihomir Culjaga 17 Mar '10

17 Mar '10

is there any way to stop/start RTP traffic on an active call? e.g. some API call or event that can be triggered through event socket ? T.

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C library and threads within openwrt
by Cavalera Claudio Luigi 16 Mar '10

16 Mar '10

Hello, I'm still working with freeswitch inside openwrt build system. What C library do you advise among these possibilities: 1) uclibc with linuxthreads-old 2) uclibc with NPTL 3) glibc (I guess this implements NPTL) I'm asking this because I know freeswitch heavy relies on threads and I've experienced some fun behaviour :-) Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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FreeSWITCH Conf Call Agenda
by Michael Collins 15 Mar '10

15 Mar '10

Hello all, The agenda for Wednesday's conference call is now up: http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_17 We could use some more content. Also, we would love to have a volunteer discuss one of the FreeSWITCH modules or some other FS feature that he/she uses. If you have any thoughts or suggestions please email me off list. Thanks! -MC

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Attended Call Transfer
by Suneel Papineni 15 Mar '10

15 Mar '10

Hi, I am trying to establish an attended call transfer scenario as follows: A call is made from extension1 to FSComm (Call 1) and is answered. Now another call is made from FSComm to extension2 and is answered (Call 2). I want to transfer the call between extension1 and extension2. For this I tried to use "uuid_bridge" command at FSComm and is issued through event sockets to FSComm as: "api uuid_bridge <A-leg Unique-id of Call1> <B-Leg Unique-id of Call 2>". This connected both the extensions and FSComm is out of loop for RTP. This achieved call transfer feature, but my issue is as follows. When I saw number of calls at FSComm (using "api show calls" command), it has shown 0 calls, that is correct. Where as when I look for number of channels (using "api show channels" command), it displayed as 2 with details of current call between two extensions (i.e. A-leg of Call 1 and B-leg of Call 2). It means FSComm didn't come out of loop completely and is occupying the two channels. Could you please let me know how can I make FSComm to come out of loop completely, so that I can accept/make another call at FSComm from/to different extension. (I tried to kill the channel at FSComm, but this is clearing the call between extensions completely) I tried to use "uuid_deflect" command, but that is useful in Blind transfer and not in attended transfer. Also tried with "uuid_transfer" command but failed to achieve. Could someone please let me know if there is any other method to achieve this attended transfer scenario. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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Offline Message Module
by Waleed 15 Mar '10

15 Mar '10

hello All i want to build application module that let FS to send IM message to the user when another user try to call him and he is Off-line or not registered now , can any one help me ? :) thanks & regards

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Skypiax has been renamed as Skypopen
by Giovanni Maruzzelli 15 Mar '10

15 Mar '10

Ciao FreeSWITCHers, is with pleasure that we're announcing the renaming of mod_skypiax module to mod_skypopen. The new wiki page is at http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk To upgrade from a previous Skypiax installation, please read: http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Upgra… Many many many thanks to the wonderful FreeSWITCH team that make it all happens! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Re: [Freeswitch-dev] JIRA Commented: (DP-15) uuid_record not working
by Mark Sobkow 14 Mar '10

14 Mar '10

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From header with 0000000000
by Suneel Papineni 12 Mar '10

12 Mar '10

Hi, This might be a basic question, but could you please let me know why I am getting "0000000000" as Caller-Caller-ID-Number (This is appearing in >From header as well) and with out From username when I am issuing following command to originate a call. api originate sofia/softphone/1004(a)172.22.252.124 9999 This could be a setting I need to make before making call. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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Problem with call recording
by Mark Sobkow 11 Mar '10

11 Mar '10

I'm having a problem with call recording. I tell Freeswitch to record a call to a file, and it responds ok, and the file shows up on the file system. I later tell Freeswitch to stop recording, and it returns ok, but the file keeps growing as it doesn't actually stop recording. As a result, any subsequent attempts to start recording the session to a different file name fails. In my case, an operator is logged in to the phone system and receiving calls from customers or from automated callbacks (if the customer didn't want to wait.) So I don't need to record the entire operator session, just _segments_ of it. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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error events for failed calls using 'conference conf_name bgdial sofia/ip/user'
by Gledhill, James 10 Mar '10

10 Mar '10

My goals are: 1) To get “start-talking” and “stop-talking” events during Early Media (before the call is answered) as well as after the call is answered. 2) If there are problems with the call, get events that tell me why the call failed. The way I am doing it accomplishes the start/stop talking (in early media) BUT if the call fails, I do not get events back letting me know it failed and why. I DO get CHANNEL_DESTROY and CHANNEL_STATE events that IF I had the UUID, I could match against, but at this point in the call I do NOT have that UUID AND when setting the UUID via “origination_uuid=my_uuid_my_uuid_my_test_uuid”, my UUID has not yet been set, so as far as I can tell there is no way to match those events with details my application has at the time. The console shows the errors (Below are two examples of calls that failed in I entered “bad” sofia locations), but they do not come back as events. 2010-03-05 14:52:20.109497 [ERR] switch_ivr_originate.c:2389 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2010-03-05 16:34:13.669509 [ERR] switch_ivr_originate.c:2389 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] So … I am wondering is there a better/different way to do it, is there a work-around, should I get events to tell me the dial had problems, etc. I am using esl/lua and esl/java to experiment with this. Both get the same results (as I would expect). I have had this as a ~take-a-number on IRC for about a week, so I thought I would add more detail and open it up to this group. I realize many of the same people do both, but hoped more eyes may look at it here and give thoughts. ===== From Pastebin ==== I am not getting 'error' events in ESL [that match a UUID I can track] when there is a problem with conference <conf_name> dial sofia/myip/1006, so I can not tell what went wrong. Is there a work-around? I start a call in a conference from ESL (conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.232/1006) If the 'dial' works (meaning does not get an error) then I get the events I want and expect. If it errors (not able to make the call ( [USER_NOT_REGISTERED] | [DESTINATION_OUT_OF_ORDER] | etc. ) then I do not get any events that tell me something went wrong. Is there a work-around for this? ===================== LUA Code run from ./freeswitch.trunk/libs/esl/lua #!/usr/local/bin/lua require("ESL") myip = "10.50.48.239" -- change this to your IP address myExt = 1006 -- change to one that is registered, then rerun with an unregistered con = ESL.ESLconnection("localhost", "8021", "ClueCon"); str = string.format("conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/%s/%s", myip, myExt) con:sendRecv("event plain ALL") ff = con:api( str ) print("** sent " .. str ) print( (ff and ff:getBody() or "no body" )) cnt = 0 while cnt <= 30 do print("loop = " .. cnt ) cnt = cnt + 1 ev = con:recvEventTimed(1000) print( ev and ev:serialize("plain") or "no event this time" ) end ============== The printout is : ** sent conference conf_uuid-TEST_CONF bgdial [origination_uuid=my_uuid_my_uuid_my_test_uuid]sofia/10.50.48.239/1006 OK Job-UUID: a5d57b34-9205-483c-ae64-60601d8178d7 loop = 0 Event-Name: CHANNEL_DESTROY Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1 FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com FreeSWITCH-IPv4: 10.50.48.232 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-03-05%2017%3A06%3A52 Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT Event-Date-Timestamp: 1267826812571550 Event-Calling-File: switch_core_session.c Event-Calling-Function: switch_core_session_perform_destroy Event-Calling-Line-Number: 1051 Channel-State: CS_NEW Channel-State-Number: 0 Channel-Name: N/A Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66 Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: ringing loop = 1 Event-Name: CHANNEL_STATE Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1 FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com FreeSWITCH-IPv4: 10.50.48.232 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-03-05%2017%3A06%3A52 Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT Event-Date-Timestamp: 1267826812571550 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_running_state Event-Calling-Line-Number: 1301 Channel-State: CS_DESTROY Channel-State-Number: 12 Channel-Name: _undef_ Unique-ID: ff9f875e-941f-438a-b384-844a6da23f66 Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: ringing loop = 2 Event-Name: API Core-UUID: 438a66f9-8e41-4c6a-bbfd-1b0e8527cec1 FreeSWITCH-Hostname: tpa-es-rh-232.dev.sipstorm.com FreeSWITCH-IPv4: 10.50.48.232 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-03-05%2017%3A06%3A52 Event-Date-GMT: Fri,%2005%20Mar%202010%2022%3A06%3A52%20GMT Event-Date-Timestamp: 1267826812571550 Event-Calling-File: switch_loadable_module.c Event-Calling-Function: switch_api_execute Event-Calling-Line-Number: 1581 API-Command: conference API-Command-Argument: conf_uuid-TEST_CONF%20bgdial%20%5Borigination_uuid%3Dmy_uuid_my_uuid_my_test_uuid%5Dsofia/10.50.48.239/1006%20 loop = 3 no event this time loop = 4

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