March 2010 - freeswitch-dev@lists.freeswitch.org

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

14 Mar '10

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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] FreeSWITCH Conference Call Agenda
by Michael Collins 10 Mar '10

10 Mar '10

Hello all! The FreeSWITCH weekly conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_10 Please feel free to add your questions and agenda items. Also, remember that this week Mathieu Rene is going to be spending a lot of time discussing mod_sofia and how it works. It should be a very informative discussion! By all means join us at 1700 GMT on Wednesday. -Michael

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[Freeswitch-dev] mod lua session:bridged() not implemented?
by rentmycoder rentmycoder 09 Mar '10

09 Mar '10

Yes you are absolutelly right, maybe I might be looking at the wrong tool... I have tried several approaches but haven't found the right one... I don't only need to detect if a call was answered, but I need also to detect and log the failure cause in case the call was not answered. Which usually can be found in the originate_disposition variable... If I use a socket command: "bgapi originate {ignore_early_media=true,originate_timeout=10,api_hangup_hook='luarun cc.lua predictive_hangup'}sofia/gateway/gwname/13246789 9999" I cannot detect the originate_disposition variable in predictive_hangup() in lua, becouse the session is nil after hangup... I think I should start dialing out from the dialplan, but in this case how to call to the dialplan using sockets? I've tried: "bgapi originate {phoneno=13246789}loopback/9999" with no success... What do you suggest? Is it possible to get this working only with 2 call legs??? I think you might be looking at the wrong tool. The session:bridged() method is for knowing if the current session is bridged to another leg. Tony says it has a very specific use case and I doubt that this is what they had in mind. If you're building a predictive dialer and you are generating the outbound call leg then you're most likely in need of knowing that the far end has answered. In a case like this I would build a dialplan extension and have the b leg drop in there. Then set execute_on_answer to your lua script. You could also use that dialplan to handle timeouts, call failures, etc. This way you're letting the dialplan do what it's good at and your Lua script can focus on the single task of handling an answered call... -MC

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[Freeswitch-dev] FSComm basic issue
by Suneel Papineni 09 Mar '10

09 Mar '10

Hi, I am trying to use FSComm with Freeswitch and facing following issues. 1. Using pre-build binary (windows), when the application is started FSComm is getting Registered properly. When I tried to make a call, UI displays Dialing... but unable to see any SIP (INVITE) messages in wireshark traces. After sometime UI displays with message "Call with (destination number) failed with reason DESTINATION_OUT_OF_ORDER though destination number is registered with another FSComm" 2. Also I am unable to see any logs generated in the log folder. Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on 26/02/10 at 4am) and tried to build FSComm. Build was succeeded. Application (FSComm) also started and displayed with UI. When I try to change the preferences, it has thrown Porta Audio Error saying "Error Querying Audio Devices" even though proper audio devices are present. Also it doesn't create folders like "conf", "mod". Even after copying all the required dll's and mod files (as specified in FSComm wiki pages), application is throwing the same error. I am using Windows XP machine. Built a Debug & Release version with 32-bit option. If someone has built FSComm for windows environment and is working fine, could you please let me know if there are any additional things I need to do to make it work. 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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[Freeswitch-dev] mod lua session:bridged() not implemented?
by rentmycoder rentmycoder 08 Mar '10

08 Mar '10

Hi Guys, Freeswitch rocks! According to the wiki page I could use (session:bridged() to check if session is bridged... http://wiki.freeswitch.org/wiki/Lua#session:bridged if (session:bridged() == true) do -- Do something end But in latest trunk session:bridged() retuns nil... If I list the session functions like this: http://wiki.freeswitch.org/wiki/Lua#How_can_I_find_useful_undocumented_Sess… session:bridged is not in the list... Do you plan to implement this function? How can I detect a bridge on the other leg in lua? I need this to create an automatic predictive dialer, which dials customers and if the customer picks up the phone the lua script plays a message and transfers the other leg to an operator... I start the call using event-socket api originate and bridges the customer with a loopback lua script: "api originate {originate_timeout=30,origination_caller_id_number=3812345678}sofia/gateway/provider/12345789 '&lua(cc.lua predictive)'" In the lua script I would have to wait for the a-leg to bridge, how to achive this without session:bridged()?

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[Freeswitch-dev] mod_skypiax (skype endpoint) call for tests
by Giovanni Maruzzelli 08 Mar '10

08 Mar '10

Hello FreeSWITCHers, it would be very useful if you could update to the latest svn and test mod_skypiax in your use cases, and report any problems. Please note that the wiki page http://wiki.freeswitch.org/wiki/Skypiax has been almost completely revamped, with complete coverage of the new options, channel variables, command line commands, events, installation, etc. Take the time to at least browse through it, because much new material has been added, that can spare you time and problems. Would be very useful if you report anything wrong, missing, or hard to understand in the wiki page itself too (btw, English is not my first language ;) ). Thanks in advance for your cooperation, that makes this community so nice a place to be in. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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[Freeswitch-dev] Call transfer using eventsockets with FSComm
by Suneel Papineni 08 Mar '10

08 Mar '10

Hi, Is there a way to transfer call with FSComm. I tried different ways using eventsockets but failed. Tried the scenario as follows: Received a call from 1001 to FSComm (registered with 1002) and is answered (Unique-id is 784dd690-6d0a-47de-b5f4-b923264581a5). Made a call from FSComm to extension 1003 and is answered (Unique-id is ec7a3bdd-c265-41fb-8cef-b897d2e8bf62). Now I want to transfer this call to 1001 and FSComm should be out of loop. a. Tried with command "api uuid_transfer 784dd690-6d0a-47de-b5f4-b923264581a5 -both park inline" but failed. With this call at 1001 is dropped and call between FSComm and 1003 is still there. b. Tried with command "api uuid_bridge 784dd690-6d0a-47de-b5f4-b923264581a5 ec7a3bdd-c265-41fb-8cef-b897d2e8bf62". With this all calls are dropped. Could someone let me know if there is any procedure for call transfer (using event sockets). 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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