May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Transcoding issue (D150 card, 1.3.4.1 library)
by Alan Scales 29 May '13

29 May '13

Hello, While setting up the environment needed to run the FreeSWITCH mod_sangoma_codec module, I've ran into a sngtc_server issue. For transcoding I'm using a D150 Sangoma transcoding card and the problem appears while starting the sngtc_server. Before starting the server, the card was configured with *sngtc_cfg*, and responded to pings. The module *detection* works, the card receives a *reset* command, and after the 20 seconds wait period, module *activation* fails with: Error: cOCTVC1_MAIN_MSG_MODULE_CLEANUP_API_RESOURCE_CID failed, rc = 0x0a000301 (cOCTVC1_GENERIC_RC_API_INVALID_CMD_LENGTH) After the failure, the procedure is repeated: { detect, reset, sleep, activate, fail } I suspect that the issue happens because of the fact that the *sngtc_server*and *card* firmwares do not match. However, after stopping the server, the firmware update (seems to be a downgrade) command also fails: Module [00-0c-90-1b-4d-6f] Upgrading Firmware cur=[01.06.02-B4-PR] to new=[01.04.03-B53-PR] Error: cOCTVC1_MAIN_MSG_FILE_OPEN_CID failed, rc = 0x0a010210 (cOCTVC1_MAIN_RC_FILE_MAX_WRITE_BYTE_SIZE) Regarding the logging, the only logs I've followed are in * /var/log/sngtc_server.log* Any hints on this matter would be highly appreciated, especially since it's not really a FreeSWITCH issue. Thank you, Alan Scales

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how to monitor a call session?
by Vincent Xia 29 May '13

29 May '13

can i monitor an ongoing call session by receiving freeswitch events through esl outbound socket? for example, i can have another process monitoring the sip call sessioin by acquiring call events(answer, hangup, etc.) from freeswitch. are the following steps ok? 1. create a socket server in a process 2. adding <action application="socket" data="IPaddr:port async"/> to the dialplan xml. 3. get the outbound socket connection from freeswitch 4. acquire call events should i do some event subscription before step 4? any response is appreciated!

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Problem with callcenter configuration (don't register agents and tiers)
by Anatolii 29 May '13

29 May '13

Hello all! Freeswitch version - 1.2.8, installed in Centos. Freeswitch always goes for configuration to our server (on which CakePHP is situated) I have problem with callcenter module in freeswitch. In mod_xml_curl add two bindings: dialplan and configuration. < binding name="configuration"> < param name="use-dynamic-url" value="yes"/> < param name="gateway-url" value="http://example.comt/freeswitch/configuration/${our_queue_name}${company_id}" bindings="configuration"/> < /binding> So, when I restart (start) freeswitch or reload mod_callcenter - FS request default configuration for callcenter and it works great (FS add queues, agents and tiers). After that in dialplan for specific number I set application callcenter with specific name and set 2 global variables, *our_queue_name* and *company_id*, which send to me with a name of queue and company id. -- this works fine When I receive from freeswitch request - I configure xml from db data and send FS callcenter configuration. -- works fine. *FS get correct xml, but add only queue settings and absolutely ignoring adding agents and tiers.* -- This is my problem for now. I try to solve this situation by using lua scripts - no results. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-callcenter-confi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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is it possible to disable hold function
by Vincent Xia 29 May '13

29 May '13

is it possible to disable hold function of a given extension by modifying freeswitch config, e.g. dialplan\default.xml?

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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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sip_ph_X- header doesn't work for freetdm-sofia call
by xiaofengcanyuexp＠163.com 28 May '13

28 May '13

Dear support I got a problem to pass through SS7 info in SIP messages. Here is the code change and test call flow: -------------------- Code change and log: -------------------- 1. At sangoma module:intercept ACM in ftmod_sangoma_ss7_*.c, parse the BCI and add it to the ftdmchan->call_data: sngss7_add_var(sngss7_info, "ss7_acm_bci_hex", val); 2. At freetdm module mod_freetdm.c, get the variable and set it to channel. Of which, the "prefix" is 'sip_ph_X-': var_value = ftdm_sigmsg_get_var(sigmsg, "ss7_acm_bci_hex"); if (!ftdm_strlen_zero(var_value)) { snprintf(variable_name, "%sTEST-BCI-HEX", prefix); switch_channel_set_variable_printf(channel, variable_name, "%s", var_value); } >From the test call log: 2013-05-27 07:03:47.747361 [INFO] ftmod_sangoma_ss7_handle.c:362 [s1c1][1:1] [CIC:1]Rx ACM 2013-05-27 07:03:47.747361 [DEBUG] ftmod_sangoma_ss7_support.c:2920 [s1c1][1:1] ACM BCI parameter Hex: 0x1636 2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5513 Get ACM BCI parameter from sangoma 2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5530 sip_ph_X-TEST-BCI-HEX:1636 --------------------- My purpose: --------------------- What I want to do is the encapsulate the sip_ph_X-TEST-BCI-HEX in the 183 SIP message. Like handling IAM in ftdm_channel_from_event(), once it sets the the variable like below, the "X-FreeTDM-SpanNumber" will automatically attached in INVITE message. switch_channel_set_variable_printf(channel, "sip_h_X-FreeTDM-SpanNumber", "%d", spanid); --------------------- Problems: --------------------- While, what's strange are: 1. I add one line in dialplan/default.xml to log the variable with: <action application="log" data="The BCI hex is: ${sip_ph_X-TRUSTID-BCI-HEX}"/> The freeswitch.log show ${sip_ph_X-TRUSTID-BCI-HEX} is empty 2. The 183 message does not attach the 'X-TRUSTID-BCI-HEX'. Can someone help to figure out what's wrong with it? Thanks ------------------- 2013-05-28

2 2

Debian build scripts - bootstrap.sh disables mod_siren?
by Alex Crow 28 May '13

28 May '13

Hi all, Just a note that in debian/ubuntu builds mod_siren has been working for quite a while (on the order of several months) for me in both squeeze and testing. Is there any reason why this is disabled? If not I will file a Jira to have it readded. Cheers Alex

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FreeSWITCH Weekly News and Notes
by Michael Collins 28 May '13

28 May '13

Happy Tuesday to you all! It is a short week for us here in the U.S. as yesterday was a national holiday. The first order of business: I am happy to let you know that Packt Publishing <http://www.packtpub.com> has informed us that the new FreeSWITCH 1.2 book <http://www.packtpub.com/freeswitch-1-2/book> is available as of May 24th! Congrats to the FreeSWITCH team for getting yet another book published. This is the third one we've done with Packt. Also, many thanks to all those who spent time answering questions and helping us with the technical reviewing process. With the new book completed we focus our attention on other things, not the least of which is ClueCon 2013 <http://www.cluecon.com>. Starting this week, those who've submitted talk proposals will be receiving the official acceptance of their presentations, including the day and time for the presentation. If you have not already submitted a talk proposal please do so right away as we have a limited number of available speaking slots. Our weekly conference calls have been lively with discussions on various topics suggested by those calling in. On last week's call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22>we had a particularly nice discussion about mod_skinny <http://wiki.freeswitch.org/wiki/Mod_skinny>. A relatively new community member, Nathan Neulinger, has really done a nice job of picking up the torch for mod_skinny. If you have a need to use Cisco phones with FreeSWITCH using SCCP<http://en.wikipedia.org/wiki/Skinny_Call_Control_Protocol>then definitely listen <http://torrents.freeswitch.org/conf_call_2013-05-22.torrent> to last week's discussion. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29> we have Martin from the VoIPMonitor.org project. We look forward to learning more about VoIPMonitor and how it can help us with keeping tabs on our VoIP servers. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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