May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

JIRA Reminder
by Brian West 06 May '14

06 May '14

FreeSWITCHers, When opening a JIRA and it asks for the Master git rev, it means one from the past 6 hours, In addition when opening a JIRA please be responsive. ‘Yes’ isn’t a valid answer, neither is a git rev from 30 days ago. Remember we are working to help you so at the very least do the leg work to help us help you. If you have any issues outstanding in JIRA please take a few moments to review them and gather the requested data or re-test with git MASTER. If no additional details can be determined please re-test you issue with Master and report those on your JIRA too. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org:81/key.txt (AB93356707C76CED)

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Phone Display during call may be CISCO SPA phone specific
by Sean Devoy 06 May '14

06 May '14

Hi, When I dial 222 from my line 220, the display says Calling To: Devoy House 222 It very quickly changes to: Session Progress To: Devoy House 222 The same is basically true calling 220 from 222 except substitute "BFIS" for "Devoy House" If I dial *9440, my dialplan transfers to extension 40 in a different realm. But the display just says: To:*9440 *9440. My question is WHERE DO THE NAMES COME FROM? I had my "internal callerid" set to "Devoy House", but nothing is set to BFIS. I changed internal caller id for that extension, reloadxml and restarted the phone. No change. This came up because a customer says some extensions get names displayed, others do not, yet another gets "Outgoing call"? There is NO Directory in the phone. I am baffled and bewildered. TIA, Sean

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Scripts directory location
by Matt Broad 06 May '14

06 May '14

Hi, is it possible to change the location of the scripts directory? I want to have one central location for multiple Freeswitch instances to read the javascript files from but have hit a snag. I have added the central location to the dialplan file when calling the javascript application e.g <action application="javascript" data="/central/freeswtich/scripts/test.js" /> This loads the initial file just fine, but any included files (include("includes/file.js")) are still trying to resolve back to /usr/local/freeswitch/scripts/includes/ directory. I have checked the java.conf.xml and spidermonkey.conf.xml files but these do not seem to hold the answer. I would prefer not to have to "hardcode" the scripts and include locations if it can be avoided. Any help would be appreciated. thanks Matt

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Unbridge Call
by Luiz Fernando Softov 06 May '14

06 May '14

Hi everyone. I have this problem making outbound call. Maybe someone can help me with this. I'm have a service in C (TCP Server), where i receive all calls. +--------+ +--------+ | A 1001 |--> make call -> dialplan -> | socket | +--------+ +--------+ <extension name="outcalls"> <condition> <action application="socket" data="127.0.0.1:8084 async full"/> <action application="hangup"/> </condition> </extension> After i accept the connection on port 8084 i send a connect command, and receive the CHANNEL_DATA event. In this service I do all the work of billing and control of active calls. I have a internal structure, of my extensions (numbers in my FS) IF EXTENSION HAVE MONEY i will send this commands to connection to make a bridge between A [1001] and B [1003] **** sendmsg\n call-command: execute\n execute-app-name: set\n execute-app-arg: dialed_extension=1003\n\n sendmsg\n call-command: execute\n execute-app-name: export\n execute-app-arg: dialed_extension=1003 sendmsg\n call-command: execute\n execute-app-name: set\n execute-app-arg: bypass_media=true\n\n sendmsg\n call-command: execute\n execute-app-name: set\n execute-app-arg: call_timeout=30\n\n sendmsg\n call-command: execute\n execute-app-name: set\n execute-app-arg: hangup_after_bridge=true\n\n sendmsg\n call-command: execute\n execute-app-name: set\n execute-app-arg: continue_on_fail=true\n\n sendmsg\n call-command: execute\n execute-app-name: bridge\n execute-app-arg: user/${dialed_extension}@${domain_name}\n\n **** - Call work fine, WITHOUT <uuid>, WITH <uuid> i receive this WARNING - [WARNING] mod_sofia.c:1281 Asynchronous PTIME not supported, changing our end from 30 to 20 The phone ring, and i answer the call, normal! But now I stopped because I not know, how to CANCEL the bridge (UNBRIDGE) and do something with the A-leg. "Something, like, HANGUP B-leg, and play a sound file to A-leg, then hangup the call." I tried, but all commands i have sended like sendmsg\n call-command: execute\n execute-app-name: hangup\n execute-app-arg: CHANNEL_UNACCEPTABLE\n\n or sendmsg\n call-command: execute\n execute-app-name: break\n\n Do nothing with the call I tried with and without "event-lock: true" I receive only this in FS log [DEBUG] switch_core_session.c:1151 Send signal sofia/internal/ 1001(a)192.168.211.69[BREAK] I tried with and without "event-lock: true" I need to do this such as: credit is over, I need to cancel the current bridge and play a sound to the user and hangup call or to simple hangup a call I receive this in FS console LOG after a-leg or bleg hangup a phone and call terminated [DEBUG] switch_ivr.c:614 sofia/internal/1001(a)192.168.211.69Command Execute hangup() [DEBUG] switch_core_session.c:2614 sofia/internal/1001(a)192.168.211.69Channel is hungup and application 'hangup' does not have the zombie_exec flag. [DEBUG] switch_ivr.c:657 sofia/internal/1001(a)192.168.211.69 skip receive message [AUDIO_SYNC] (channel is hungup already) This is for each "sendmsg" i send after bridge was answered. Sorry about my english...

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One way audio only in a specified context
by Mr.Gab 06 May '14

06 May '14

Hello everybody, I have a problem with a FS installed on the cloud (configured following Freeswitch EC2 Wiki). Everything is ok except a one way audio behaviour that i have only if i transfer an outbound call. If I transfer an inbound call I don't have this problem. I'll explain the scenario with a little schema INBOUND CALL -> Extension 100 -> transfer to 101 WORKS. Extension 100 -> OUTBOUND CALL -> transfer to 101 ONE WAY AUDIO. I have one snom 300 (extension 100) and 3 Gigaset A510IP behind a tomato router. Can someone suggest me some way to manage this issue? Thanks in advice crazygabry

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dial plan in mod xml curl
by Shaik Raheemuddin 06 May '14

06 May '14

Hi, I am trying to use mod xml curl with freeswitch. i have configured in "xml_curl.conf.xml " gateway url to <param name="gateway-url" value="http://myipadress/freeswitchdata" bindings="dialplan|configuration|directory"/> where in at server it sends below response <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition></extension></context></section> <section name="directory"> <domain name="123.176.38.68"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <groups> <group name="default"> <users> <user id="1234" cacheable="true"> <params> <param name="reverse-auth-user" value="1234" /> <param name="reverse-auth-pass" value="1234" /> </params> </user> <user id="4321" cacheable="true"> <params><param name="reverse-auth-user" value="4321" /> <param name="reverse-auth-pass" value="4321" /> </params> </user> </users> </group> </groups> </domain> </section> </document> I am unable to make call for below error 2014-05-06 12:44:15.313984 [INFO] mod_dialplan_xml.c:558 Processing 1234 <1234>->4321 in context public 2014-05-06 12:44:15.633985 [INFO] switch_core_state_machine.c:192 No Route, AbortingHi, I am trying to use mod xml curl with freeswitch. i have configured in "xml_curl.conf.xml " gateway url to <param name="gateway-url" value="http://myipadress/freeswitchdata" bindings="dialplan|configuration|directory"/> where in at server it sends below response <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition></extension></context></section> <section name="directory"> <domain name="123.176.38.68"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <groups> <group name="default"> <users> <user id="1234" cacheable="true"> <params> <param name="reverse-auth-user" value="1234" /> <param name="reverse-auth-pass" value="1234" /> </params> </user> <user id="4321" cacheable="true"> <params><param name="reverse-auth-user" value="4321" /> <param name="reverse-auth-pass" value="4321" /> </params> </user> </users> </group> </groups> </domain> </section> </document> I am unable to make call for below error 2014-05-06 12:44:15.313984 [INFO] mod_dialplan_xml.c:558 Processing 1234 <1234>->4321 in context public 2014-05-06 12:44:15.633985 [INFO] switch_core_state_machine.c:192 No Route, Aborting -- Thanks and Regards Shaik Raheemuddin

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Hook on SIP CANCEL
by Oleg Stolyar 06 May '14

06 May '14

Hi guys, I am using the api_hangup_hook variable to let my internal system know when a call is terminated. Every now and then my calling party cancels the SIP Invite before my dial plan is finished. IN this case FreeSWITCH properly responds with a "487 Request Terminated" However, this does not fire the command assigned to the api_hangup_hook variable. This is probably also proper since the call has not yet been established. My question is whether there is a place somewhere where I can hook into the processing of the CANCEL SIP message. Thank you Oleg

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Cancell Bridge
by Luiz Fernando Softov 06 May '14

06 May '14

Hi everyone. I have this problem making outbound call. Maybe someone can help me with this. I'm have a service in C (TCP Server), where i receive all calls. <extension name="outcalls"> <condition> <action application="socket" data="127.0.0.1:8084 async full"/> <action application="hangup"/> </condition> </extension> In this service I do all the work of billing and control of active calls. This is my commands, where i receive a connection to make a call to 1001 sendmsg execute\n call-command: set\n execute-app-name: dialed_extension=1001\n\n sendmsg execute\n call-command: export\n execute-app-name: dialed_extension=1001 sendmsg execute\n call-command: set\n execute-app-name: bypass_media=true\n\n sendmsg execute\n call-command: set\n execute-app-name: call_timeout=30\n\n sendmsg execute\n call-command: set\n execute-app-name: hangup_after_bridge=true\n\n sendmsg execute\n call-command: set\n execute-app-name: continue_on_fail=true\n\n sendmsg execute\n call-command: bridge\n execute-app-name: user/${dialed_extension}@${domain_name}\n\n But now I stopped because I can not cancel the bridge and do something with the call. I tried, but all commands i have sended like sendmsg execute\n call-command: hangup\n execute-app-name: CHANNEL_UNACCEPTABLE\n\n or sendmsg execute\n call-command: break\n\n Do nothing with the call I receive only this in FS log [DEBUG] switch_core_session.c:1151 Send signal sofia/internal/ 1001(a)192.168.211.69[BREAK] I tried with and without "event-lock: true" I need to do this such as: credit is over, I need to cancel the current bridge and play a sound to the user and hangup call or to simple hangup a call I receive this in FS console LOG after a-leg or bleg hangup a phone and call terminated [DEBUG] switch_ivr.c:614 sofia/internal/1001(a)192.168.211.69Command Execute hangup() [DEBUG] switch_core_session.c:2614 sofia/internal/1001(a)192.168.211.69Channel is hungup and application 'hangup' does not have the zombie_exec flag. [DEBUG] switch_ivr.c:657 sofia/internal/1001(a)192.168.211.69 skip receive message [AUDIO_SYNC] (channel is hungup already) This is for each "sendmsg" i send after bridge was answered.

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ClueCon Weekly - May 7 TJ Chats about TMUX
by Ken Rice 05 May '14

05 May '14

Greetings! This week on ClueCon Weekly, TJ aka Bougyman is going to join us for a tmux demo and show you another tool to help administer servers without multiple SSH sessions rolling along. Join us Wed May 7 at 1P Eastern (1700 GMT) on sip:888@conference.freeswitch.org ! Call starts promptly at the top of the hour, so get in a little early and have a chat with the crew before we get started. Don¹t have time before? Hang around for the after party and get your FS questions answered. Coming up May 14th Giovanni Maruzzelli will be joining us to talk about linking FreeSWITCH to GSM networks via GSMOpen And don¹t forget to increase your Chances of winning @ ClueCon https://www.cluecon.com/increases-your-chances-of-winning/ --K -- Ken https://www.cluecon.com/increases-your-chances-of-winning/ http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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ClueCon 2014 - Register Now Increases Your Chances of Winning
by Ken Rice 05 May '14

05 May '14

<https://cluecon.com>[image: https://billing.freeswitch.org/cart.php?gid=1]<https://billing.freeswitch.org/cart.php?gid=1> *ClueCon 2013 <https://cluecon.com/> - Register Today! <https://billing.freeswitch.org/cart.php?gid=1> * At ClueCon we love giving stuff away. Our sponsors have given us many cool toys to give away this year. We’ll have things from ThinkGeek, Snom, Packt Publishing and others! You can increase your chances of winning by registering right away. Everyone who registers by Friday May 16th will get 16 extra drawing tickets! You can register online right now. Just visit the ClueCon.com<http://cluecon.com/> or click Register Now <https://billing.freeswitch.org/cart.php?gid=1>. Great Presentations: WebRTC, VoIP Security, and Much Much More! Need help registering for cluecon? Operators are standing by! 1 877-742-CLUE The Schedule for speakers is starting to fill out! If you want to present at ClueCon this year, email your speaking proposal to marketing(a)cluecon.com! Feel free to contact us if you have any questions about this year's event. We look forward to seeing everyone in Chicago this August!

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