June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

LUA BUG not updating the hangup cause once originated the call
by Ashou Han 30 Jun '14

30 Jun '14

Hi there, There is a bug for Lua, that the hangup cause not updated once originated the call. I originated a call to my opensips proxy with a 183 back, lua hangupCause return SUCCESS, but few seconds later the far end carrier return USER_BUSY (see this in the DEBUG log). But session:hangupCause() still return SUCCESS, not the updated cause: USER_BUSY. FS version: FreeSWITCH Version 1.2.22+git~20140309T154101Z~455941d369~64bit (git 455941d 2014-03-09 15:41:01Z 64bit) FreeSWITCH Version 1.2.23+git~20140506T190611Z~88216387e0~64bit (git 8821638 2014-05-06 19:06:11Z 64bit) FreeSWITCH Version 1.2.10+git~20130628T211259Z~e29731d673 (git e29731d 2013-06-28 21:12:59Z) All have the same bug! Thanks, Ashou Han

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Multiple inbound ACL on external profile
by Joel White 30 Jun '14

30 Jun '14

Is it possible to multiple inbound acl lists on a single profile?

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set variable on first leg when bridge is answered
by Grant Bagdasarian 30 Jun '14

30 Jun '14

Hello, I'm first doing a originate which hits the below dialplan when answered. When it's answered I'm setting a variable named "bridged" to 0. When the user presses the "critical_option_digit" the call is bridged to another destination. I need to know if this bridge was answered or not. I tried different ways of execute_on_answer but that didn't work: <action application="set" data="execute_on_answer=set bridged=1"/> <action application="set" data="nolocal:execute_on_answer=set bridged=1"/> <action application="export" data="execute_on_answer=set bridged=1"/> <action application="export" data="nolocal:execute_on_answer=set bridged=1"/> It is important the bridged variable is set to 1 on the first leg and a header is appended to the BYE/200 OK(BYE) of the first leg containing this new value. Any ideas how to fix this? <?xml version="1.0" encoding="utf-8"?> <include> <context name="outbound_cc"> <extension name="start"> <condition field="destination_number" expression="^(start)$" require-nested="false"> <action application="answer" /> <action application="set" data="bridge_early_media=true" /> <action application="set" data="hangup_after_bridge=false" /> <action application="set" data="critical_option_digit=1" /> <action application="set" data="transfer_destination=0031123456789" /> <action application="set" data="bridged=0" />  <action application="play_and_get_digits" data="1 1 2 5000 # welcome.wav invalid.wav dtmfInput \d 2500"/> <action application="transfer" data="act_on_input XML outbound_cc" /> </condition> </extension> <extension name="act_on_input"> <condition field="${dtmfInput}" expression="${critical_option_digit}" break="never">  <action application="set" data="execute_on_answer=set bridged=1"/> <action application="bridge" data="{sip_copy_custom_headers=false}sofia/outbound/${transfer_destination}@$${outbound_load_balancer_sip_ip}:$${outbound_load_balancer_sip_port}" /> <action application="export" data="sip_bye_h_X-CollectedData=<CollectedData><DTMFInput>${dtmfInput}</DTMFInput><BridgeSucceeded>${bridged}</BridgeSucceeded></CollectedData>"/> <action application="export" data="sip_rh_X-CollectedData=<CollectedData><DTMFInput>${dtmfInput}</DTMFInput><BridgeSucceeded>${bridged}</BridgeSucceeded></CollectedData>"/>  <anit-action application="export" data="sip_bye_h_X-CollectedData=<CollectedData><DTMFInput>${dtmfInput}</DTMFInput><BridgeSucceeded>${bridged}</BridgeSucceeded></CollectedData>"/> <anit-action application="export" data="sip_rh_X-CollectedData=<CollectedData><DTMFInput>${dtmfInput}</DTMFInput><BridgeSucceeded>${bridged}</BridgeSucceeded></CollectedData>"/> <anti-action application="hangup"/> </condition> </extension> </context> </include>

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failed to play audio
by Joel Dodson 30 Jun '14

30 Jun '14

Hi, I'm having intermittent trouble playing audio in a call. The scenario is: - there's an incoming call which triggers an outgoing call. - Audio is played to the incoming leg while the outgoing leg is ringing. - Once the outgoing call is connected, audio on the incoming legs is stopped and the legs are bridged - recording is started on the inbound call leg - audio is being played on the outgoing leg while the legs are being bridged. This is all done using event sockets, not any of the advanced features of the XML dialing plan. The problem is intermittent and hard to reproduce (of course, it's easily reproducible in our production environment by a customer...). I have found a significant difference in logs on calls where the problem has occurred. On the problem call, I don't see a call made to mod_httapi to cache the audio file to be played. Here are some logs for a call with the problem (newer logs are at the top, I extracted these logs from a logging aggregation tool) : 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:483 (c-4fkbyrefwy3jgl2e5kfzori) State SOFT_EXECUTE" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:415 (c-4fkbyrefwy3jgl2e5kfzori) Running State Change CS_SOFT_EXECUTE" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:473 (c-4fkbyrefwy3jgl2e5kfzori) State RESET going to sleep" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_session.c:1333 Send signal c-4fkbyrefwy3jgl2e5kfzori [BREAK]" 2014-06-26 19:57:59.604394 [DEBUG] switch_ivr_bridge.c:764 (c-4fkbyrefwy3jgl2e5kfzori) State Change CS_RESET -> CS_SOFT_EXECUTE"2014-06-26 19:57:59.604394 [DEBUG] switch_ivr_bridge.c:757 c-4fkbyrefwy3jgl2e5kfzori CUSTOM RESET" 2014-06-26 19:57:59.604394 [DEBUG] mod_sofia.c:168 c-4fkbyrefwy3jgl2e5kfzori SOFIA RESET" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:473 (c-4fkbyrefwy3jgl2e5kfzori) State RESET" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:415 (c-4fkbyrefwy3jgl2e5kfzori) Running State Change CS_RESET" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:492 (c-4fkbyrefwy3jgl2e5kfzori) State HIBERNATE going to sleep" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_session.c:1333 Send signal c-4fkbyrefwy3jgl2e5kfzori [BREAK]" 2014-06-26 19:57:59.604394 [DEBUG] switch_ivr_bridge.c:772 (c-4fkbyrefwy3jgl2e5kfzori) State Change CS_HIBERNATE -> CS_RESET" 2014-06-26 19:57:59.604394 [DEBUG] mod_sofia.c:224 c-4fkbyrefwy3jgl2e5kfzori SOFIA HIBERNATE" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:492 (c-4fkbyrefwy3jgl2e5kfzori) State HIBERNATE" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:415 (c-4fkbyrefwy3jgl2e5kfzori) Running State Change CS_HIBERNATE" 2014-06-26 19:57:59.604394 [DEBUG] switch_core_state_machine.c:477 (c-4fkbyrefwy3jgl2e5kfzori) State EXECUTE going to sleep"2014-06-26 19:57:59.604394 [DEBUG] switch_ivr.c:612 c-4fkbyrefwy3jgl2e5kfzori Command Execute playback({tagId=tag-6gtqkkwg62k7xziefiyv7vq,}http://(nohead=true,ext=mp3)www.t tsprovider.com/tts/gen.php?EID=2&LID=1&VID=1&TXT=Call+from+Direct&EXT=mp3&A…) EXECUTE c-4fkbyrefwy3jgl2e5kfzori playback({ta gId=tag-6gtqkkwg62k7xziefiyv7vq,}http://(nohead=true,ext=mp3) www.ttsprovider.com/tts/gen.php?EID=2&LID=1&VID=1&TXT=Call+from+Direct&EXT=… CS=abcde0123456789)" 2014-06-26 19:57:59.534396 [DEBUG] sofia.c:7541 dispatched freeswitch event for INFO" 2014-06-26 19:57:59.534396 [DEBUG] switch_core_session.c:915 Send signal c-4fkbyrefwy3jgl2e5kfzori [BREAK]" and here's what we see in the general case (these logs are directly from freeswitch.log so newest logs at the top): 2014-06-28 21:22:13.086522 [DEBUG] switch_ivr.c:612 c-dvji55mpysr3gcyruol6fgi Command Execute playback(http:// (nohead=true,ext=mp3) www.ttsprovider.com/tts/gen.php?EID=2&LID=1&VID=1&TXT=Hi+guys&EXT=mp3&ACC=X… ) EXECUTE c-dvji55mpysr3gcyruol6fgi playback(http://(nohead=true,ext=mp3) www.ttsprovider.com/tts/gen.php?EID=2&LID=1&VID=1&TXT=Hi+guys&EXT=mp3&ACC=X… ) 2014-06-28 21:22:13.086522 [DEBUG] switch_event.c:1608 Parsing variable [nohead]=[true] 2014-06-28 21:22:13.086522 [DEBUG] switch_event.c:1608 Parsing variable [ext]=[mp3] 2014-06-28 21:22:13.446522 [DEBUG] mod_httapi.c:2482 caching: url: http://www.ttsprovider.com/tts/gen.php?EID=2&LID=1&VID=1&TXT=Hi+guys&EXT=mp… to /var/lib/freeswitch/storage/http_file_cache/e0aaab89166c3ebf4391ec147a1d0b2a.mp3 (8198 bytes) 2014-06-28 21:22:13.446522 [DEBUG] switch_ivr_play_say.c:1308 Codec Activated L16@8000hz 1 channels 20ms 2014-06-28 21:22:14.546521 [DEBUG] switch_ivr_play_say.c:1693 done playing file http://(nohead=true,ext=mp3) www.ttsprovider.com/tts/gen.php?EID=2&LID=1&VID=1&TXT=Hi+Avery+guys&EXT=mp3… As you can see, in the error case, instead of caching the the audio from our tts provider, the channel transitions immediately into HIBERNATE state. We're using the Java Event Socket Client ( https://wiki.freeswitch.org/wiki/Java_ESL_Client) to send/receive commands and events to/from freeswitch. We're using version 0.9.2 directly from the maven repository. And here's our freeswitch version: FreeSWITCH version: 1.2.8+git~20130430T234615Z~95d31525ce (git 95d3152 2013-04-30 23:46:15Z) I've been digging into this for quite some time. I'm concerned we're doing a lot of stuff in parallel that might be causing problems internally in freeswitch. I'd be very grateful for any suggestions on what's happening and how we can avoid it. thanks, Joel

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Disable ipv6 altogether?
by Doug Vought 30 Jun '14

30 Jun '14

Is there a way to disable a gateway from ipv6 altogether? I'm having problems with registration that I think may be caused by multiple ipv4/ipv6 registrations to the same host.

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gsmopen remote hangup/no answer detection issue
by Dmitry Sytchev 30 Jun '14

30 Jun '14

When I originate call with gsmopen and Huawei e169 dongle, call gets delivered to callee, and if he drops call, modem doesn't get hangup notification or busy indication. There are no difference whether call was answered or not, when calle hangs up, modem on our side continues to ring or just to send media from other side. Maybe there should be some special AT commands or something operator-specific to make this work? There is no messages from modem in logs except these after originate timeout hangup. I haven't found any issues like this in google and this list, so maybe its just our misconfiguration... Should I file a JIRA? -- Best regards, Dmitry Sytchev, IT Engineer

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Re: [Freeswitch-users] condition
by aademattia＠comcast.net 28 Jun '14

28 Jun '14

I am trying to find out if the variable is not set. The problem is this condition fires if the variable is set or not set. If the variable is not set, I want it to sleep. Sent from my HTC ----- Reply message ----- From: "Ken Rice" <krice(a)freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] condition Date: Sat, Jun 28, 2014 1:24 PM thats tesing for an empty string... test to see if what you expect to be there and if not use anti-action to sleep Ken Sent from my iPad On Jun 28, 2014, at 11:15, "Andrew" <aademattia(a)comcast.net> wrote: Hi,I want to see if the variable is set. If its not I want to sleep for 500 mill sec.This does not seem to work. <condition field="${my_dialstring}" expression="^$"><action application="sleep" data="500"/> </condition> _________________________________________________________________________ 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

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condition
by Andrew 28 Jun '14

28 Jun '14

Hi, I want to see if the variable is set. If its not I want to sleep for 500 mill sec. This does not seem to work. <condition field="${my_dialstring}" expression="^$"> <action application="sleep" data="500"/> </condition>

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Freeswitch + gsmopen outgoing call problem
by Sangram Rath 27 Jun '14

27 Jun '14

Hi, Have a setup of freeswitch + gsmopen, incoming call works fine. However when I dial out it does not do anything. What I am trying to do here is connect 24 huawei e169 modems and have 12 of them call the other 12. The modem is able to get incoming but cannot dial out either to another number or another modem. modems are all unlocked. I have configured the opengsm.conf.xml file and the devices are detected and work well. I am new to this, so please let me know what info is needed from me like logs and config file. -- Thanks, Sangram Rath

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 27 Jun '14

27 Jun '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday BugHunt Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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