April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

voicemail_say_phone_number
by Yuriy Nasida 24 Jun '13

24 Jun '13

Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise.Thanks.

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A plea from your lead developer: Help my kid play in a Major League Ballpark!
by Anthony Minessale 21 Jun '13

21 Jun '13

Hello, My son is an aspiring baseball player on a select team here in Wisconsin. His team, The Wisconsin Wildcats, has a really special chance to get to play a game inside Miller Park. This is the Major League park where the Milwaukee Brewers play and not very easy for a 13yr old to make it to. The team has to sell as many tickets as possible to 2 games happening in April and May to get the opportunity to play. Everyone on the team is trying hard to sell the tickets and so am I. One problem is most of the people I know live far away =D So, if you do live anywhere near the Milwaukee area and like baseball, the games are: Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. I will include a FREE copy of FreeSWITCH with any ticket purchase or donation! If you live close enough to attend one of these games or will be in the area, email me offline and i can get you the other details. If you live far away and still want to help, send paypal donation to brewers(a)freeswitch.org or to the one on our site with some mention of BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf and give them to worthy local baseball fans. Here's a unique chance to thank my son for sharing his dad's time with all of you out there using FreeSWITCH! There is not much time to get all the tickets sold so if you can help, act now! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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Anyone able to get freeswitch to register a gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
by Mitch Capper 14 Jun '13

14 Jun '13

I am trying to get freeswitch to register with another freeswitch server (on TLS port 5061) however default the config settings I believe should set it, it always trys to register over UDP (after registration or with registration set to false however it does send TCP then for outbound calls). If anyone has a gateway config that actually registers over TCP or TLS that would be great. The client gateway/profile config at: http://pastebin.freeswitch.org/15383is what i believe should work but isn't I have tried setting contact-params on the gateway to tport=tls tport=tcp tport=tcp;transport=tls and transport=tls without any luck. Anyway even TCP would be helpful as it would allow me to trace better how the sip registration decides to use TCP or UDP. I tried digging into libsofia but its a bit hard to trace and understand with its generic event structure and tagging (plus the documentation isn't the most rebust on registration options). Thanks ~Mitch

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Kamailio+Freeswitch SBC
by Camila Troncoso 14 Jun '13

14 Jun '13

Hi all, I’m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * When I make a call, from one client coneccted to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing thourgh a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , becouse the second gateway is also in the cloud, but when this happens, Fresswitch answer with : “482 Request merged”. Is there a way to solve this? Regards, * * *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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USSD support for mod_gsmopen
by Yuriy Ostapchuk 11 Jun '13

11 Jun '13

Hi, I've submitted a patch that implements USSD messaging support. (useful for checking and refilling account balance and other provider-specific usage) Here is the JIRA ticket: http://jira.freeswitch.org/browse/FS-5078 If this feature is considered useful, I would appreciate if it's integrated in git. Thanks, Yuriy

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How to add username in SIP packet OPTIONS
by "Гришин В.С." 11 Jun '13

11 Jun '13

My gataway profile <include> <gateway name="SIP_M200"> <param name="username" value="USERNAME_STRING"/> <param name="realm" value="172.30.0.2"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> </gateway> </include> SIP packet which is accepted by the remote UA Thu Nov 29 16:38:24 2012 RECEIVE 15:5060 <-- 172.30.0.1:5080 OPTIONS sip:172.30.0.2;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bK6DKm54N817tUS Max-Forwards: 70 *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ To: <sip:172.30.0.2>* Call-ID: 79818268-b4c4-1230-60ac-00d0b7ded66a CSeq: 36758947 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 but in reply arrives nothing... the developer points to that that in a SIP packet there is no field username in *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ <----in this string no username To: <sip:172.30.0.2> <---- and in this string no username* it is necessary to make as From: "USERNAME_STRING" <sip:USERNAME_STRING@172.30.0.1>;tag=9DcB7gSNZZ41j What it is necessary to change or add in a gateway profile (or other place) for this purpose? Vladislav Grishin

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Setting up callbacks
by Alex Lake 07 Jun '13

07 Jun '13

I was wondering where's a good place to find some examples of how, by sending the right commands to the event_socket, I could have Freeswitch establish callbacks for me. Essentially there are a couple of different termination types - PSTN (via a gateway) and internally registered SIP accounts I would wish to be able to set up A->B (and maybe small conferences) using these types of destination in any combination. I've looked through the event_socket pages and the "originate" syntax, but would like to put together something a little more "idiot-friendly" so am looking around for precedents/tips...

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IVR EXTERNAL TIMEOUT BRIDGE PROBLEM
by Diego Mendieta 29 May '13

29 May '13

From: d_iego(a)msn.com To: freeswitch-users-owner(a)lists.freeswitch.org Subject: IVR EXTERNAL TIMEOUT BRIDGE PROBLEM Date: Thu, 11 Apr 2013 18:08:51 +0000 Hello World! I've been trying to solve an issue with my freeswitch pbx, please help me. I have set up a Welcome IVR for all incoming calls. Its seems to work perfectley when dialed from internal extentions. The problems is that when I dial from an external phone the Exit Action is not executed, instead the phone call is terminated. I am using fusionpbx. Below I paste the log for both cases: EXECUTE sofia/external/76xxxx99(a)200.119.223.228 sleep(2000) 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal sofia/external/76xxxx99(a)200.119.223.228 [BREAK] 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal sofia/external/76xxxx99(a)200.119.223.228 [BREAK] 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal sofia/external/76xxxx99(a)200.119.223.228 [BREAK] 2013-04-11 13:49:11.749600 [DEBUG] sofia.c:5431 Channel sofia/external/76xxxx99(a)200.119.223.228 entering state [ready][200] EXECUTE sofia/external/76xxxx99(a)200.119.223.228 ivr(bienvenida) 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exit' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-sub' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exec-app' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-play-sound' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-back' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-top' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:796 building menu 'bienvenida' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '1' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '2' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '3' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '/(^\d{3,6}$)/' 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:433 Executing IVR menu bienvenida 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@8000hz 1 channels 20ms 2013-04-11 13:49:13.549600 [DEBUG] switch_rtp.c:3204 Correct ip/port confirmed. 2013-04-11 13:49:29.749600 [DEBUG] switch_ivr_play_say.c:1672 done playing file C:\Program Files\FusionPBX\sounds/es/mx/maria/ivr/bienvenida.wav 2013-04-11 13:49:29.749600 [DEBUG] switch_ivr_menu.c:348 waiting for 5/5 digits t/o 2000 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:395 digits '' 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:446 Maximum timeouts 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:599 exit-sound '(null)' 2013-04-11 13:49:29.769600 [NOTICE] switch_core_state_machine.c:226 sofia/external/76xxxx99(a)200.119.223.228 has executed the last dialplan instruction, hanging up. 2013-04-11 13:49:29.769600 [DEBUG] switch_channel.c:2846 (sofia/external/76xxxx99(a)200.119.223.228) Callstate Change ACTIVE -> HANGUP internal call:EXECUTE sofia/internal/101@192.168.0.26:5060 sleep(1000) EXECUTE sofia/internal/101@192.168.0.26:5060 set(hangup_after_bridge=true) 2013-04-11 14:01:38.948600 [DEBUG] mod_dptools.c:1286 sofia/internal/101@192.168.0.26:5060 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/101@192.168.0.26:5060 ivr(bienvenida) 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exit' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-sub' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exec-app' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-play-sound' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-back' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-top' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:796 building menu 'bienvenida' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '1' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '2' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '3' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '8' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '9' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '/(^\d{3,6}$)/' 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:433 Executing IVR menu bienvenida 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@8000hz 1 channels 20ms 2013-04-11 14:01:55.188600 [DEBUG] switch_ivr_play_say.c:1672 done playing file C:\Program Files\FusionPBX\sounds/es/mx/maria/ivr/bienvenida.wav 2013-04-11 14:01:55.188600 [DEBUG] switch_ivr_menu.c:348 waiting for 5/5 digits t/o 2000 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:395 digits '' 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:446 Maximum timeouts 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:599 exit-sound '(null)' EXECUTE sofia/internal/101@192.168.0.26:5060 bridge(sofia/internal/1001(a)192.168.0.26) 2013-04-11 14:01:55.208600 [DEBUG] switch_channel.c:1045 sofia/internal/101@192.168.0.26:5060 EXPORTING[export_vars] [domain_name]=[192.168.0.26] to event 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2013-04-11 14:01:55.208600 [NOTICE] switch_channel.c:924 New Channel sofia/internal/1001(a)192.168.0.26 [c5904e71-b9ab-4f60-a098-9c8280f38aa7] Below I paste my dial plan configuration: <extension name="bienvenida" > <condition field="destination_number" expression="^7000$" > <action application="answer" /> <action application="sleep" data="1000" /> <action application="set" data="hangup_after_bridge=true" /> <action application="ivr" data="bienvenida" /><action application="bridge" data="sofia/internal/1001(a)192.168.0.26"/> </condition></extension> Please help me. thank you, Diego

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Best Installation MOD_GSMOPEN
by Josue Diaz Cruz 29 May '13

29 May '13

Good Afternoon. I beg your pardon if my question is really from a newbee. I am trying to install gsmopen in a system. I did it with CentOS 6 X64, Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. Cause i allways have problems in one point of the installation. Depending the distro could it be one or other. I was thinking that could be nice to give a complete requierements (libs) to install before do anything. If some one can provide me a easy, simple and not troubles, could be nice: * System and version * List of requierements. * Better steps to avoid issues with libraries and compile. What i tried: Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when try to load mod_gsmopen.so) Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) E169 works perfectly. I have a mobile partner with voicecall and i test it calling my land line without issues. Thank you very much for your consideration.

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FreeSWITCH 1.0.X.... Really people?
by Ken Rice 29 May '13

29 May '13

So I have noticed lately that tons of people are still trying to install and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a *VERY* bad idea... These old versions of FreeSWITCH have known vulnerabilities in them and are no longer supported by FreeSWITCH.org period. The 1.2 branch has been released for quite a while now, and is making excellent progress in the terms of both stability and reliability. So now, if you post a Jira for help on FreeSWITCH 1.0 the response you will get a closed bug with the comments to update and try again. If you post to the mailing list you will get told to update and try again If you ask on irc you¹ll get told to update and try again (after getting laughed at and made fun of) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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