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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

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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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phone hangs when placing internal calls
by Chris Brannon 28 May '13

28 May '13

Hello, I'm very new to freeswitch and telephony in general. I've managed to register phones with my server, but when I place test calls, the phone hangs, even though fs_cli tells me that the call was completed successfully. Also, everything looks fine when I run sip trace. FWIW, I've tried both linphone on Linux and csipsimple on Android. All of this is taking place on my LAN, so NAT isn't an issue. I started looking at traffic with tcpdump, and here's what I found. The phone authenticates to FreeSwitch and initiates the call. FreeSwitch sends a "100 trying" SIP message, followed by a UDP packet containing what appears to be SDP. It has no SIP header. This is followed by RTP from FreeSwitch to the phone. There is never a "200 OK" SIP message. Next, I ran linphone in debugging mode. I'll include a link to the log, in the hope that it will be helpful. Apparently, linphone isn't parsing the packet after the 100 trying message. Any idea what is going on here? http://the-brannons.com/linphone.log -- Chris

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voicemail is not working
by juned 28 May '13

28 May '13

Hi All, I am newbie to FS. so as a startup i have installed FS in mu local system to test out the basic functionality and features. so i have registered default users 1000 and 1001 in softphone( twinkle <http://mfnboer.home.xs4all.nl/twinkle/> ). registration was successful and calls was also fine but when i tried to check voicemail then it didn't worked. what i did to test it out voicemail is, did call to 1001 and let it rang so after 30 second if no answer is there then voicemail will be activated but calls are released after 30 seconds. As per documentation i came to know that by default voicemail is activated in default extensions is it so ? or i am missing something. Please point me to right direction, i want to have such a dialplan in which user can leave voicemail in cases of busy,unavailable and not answering. Thanks & Regards Juned -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/voicemail-is-not-working-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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ESL using bridge app doesn't return which gateway was used
by Clinton Goudie-Nice 28 May '13

28 May '13

When you make a bridge command using esl, where you specify multiple gateways or sip dials separated by or bars, you can't figure out which gateway was used. For example, if you bridge to something like this: sofia/gateway/SBC-GW2/+18019600000|sofia/gateway/SBC-GW1/+18019600000 The call could be bridged to either GW2 or GW1. When the CHANNEL_BRIDGE event is returned, you can see the original string in variable_current_application_data, and you may be able to infer the destination based on IP address, but nothing clearly says what gateway is used. If you turn on the all events firehose, you can see the CHANNEL_CREATE event come over the socket, and it does contain variable_sip_gateway_name with the actual name of the gateway, however I can't devise a way to access that data using the org.freeswitch.esl.client library, and even if I could, I still don't want all events for this system. Is it possible to get this information returned in any meaningful way through the ESL layer, either by an api command to query, or the setting of a variable that will give me back which gateway a bridge was performed through? If none of that is possible, this sounds worthy of filing a bug to return the variable_sip_gateway_name in the CHANNEL_BRIDGE event. Thanks for the help, Clint

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add-users scripts
by Adam Cooley 28 May '13

28 May '13

hey gang, new to freeswitch here..I installed freeswitch 1.2.9 via centos, with yum, and was wondering where I would procure the add-user scripts for adding users on the cli? i can't seem to find them on the filesystem.. -adam

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