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Segfault on startup when mod_freetdm is enabled.
by Eric Michel 04 Mar '11

04 Mar '11

I updated to the latest git last night, and started having issues. I assumed I'd screwed up and (not being very adept at unistalling things from source) decided to do a clean install today (including the OS). When I enable mod_freetdm in modules.conf.xml, freeswitch segfaults on startup. If I go back and disable mod_freetdm it starts fine. As this is a clean install the only thing I've changed is modules.conf.xml. Is any one else having this problem? Do I need to file a bug report or am I just missing something really simple? I'm running FreeSWITCH Version 1.0.head(git-64806d2 2011-03-02 18-23-19 +0100). I installed the latest non-beta version of Wanpipe which is 3.5.18, and used "make freetdm" when I compiled it. I've got the Sagnoma A200 card. The OS is CentOS 5.5 64bit, AMD Athalon 7850 Dual-core processor, 2Gb RAM, 2 500Gb HD in a software RAID1. Before I compiled freeswitch I enabled "../../libs/freetdm/mod_freetdm" and "asr_tts/mod_flite" in modules.conf. I've run wancfg_fs. Eric

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LUA FS's API function "freeswitch.email" do not work
by Dmitry Saratsky 04 Mar '11

04 Mar '11

Hi all, According to: http://wiki.freeswitch.org/wiki/Lua#freeswitch.email I should successfully send emails from within FS with following script: 19:19 fs:/usr/local/freeswitch/scripts# cat test.lua freeswitch.email("user(a)domain.com", "user(a)domain.com", "subject: Voicemail from 1234\n", "Hello,\n\nYou've got a voicemail, click the attachment to listen to it.") 19:19 fs:/usr/local/freeswitch/scripts# When I run script, I can see in console: freeswitch@default> lua test.lua -ERR no reply 2011-03-04 19:20:46.783430 [DEBUG] switch_utils.c:709 Emailed data to [user(a)domain.com] freeswitch@default> I have the following configured in my autoload_configs/switch.conf.xml: <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-t"/> However nothing passed to sendmail... it is no records in /var/log/maillog. I run: CentOS release 5.5 (Final) FreeSWITCH Version 1.0.head (git-53fc3f7 2011-02-28 12-43-05 -0600) Any ideas?

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Dialing extensions behind NAT
by Ali Zaidi 04 Mar '11

04 Mar '11

Hi, Not sure if this was previously asked, my scenario as following. UA1,UA2,UA3===(NAT) ====(Internet)====(FreeSwitch)===(PSTN) When UA1 dial UA2 the traffic go over the internet connection to FreeSwitch and then comes back to UA2. Can UA1 dial UA2 without going to internet? My both phones are Linksys SPA942, please let me know any good phones that can work best with free switch over wan connection. Thanks, AGZ

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mod_say and bargein
by Tihomir Culjaga 04 Mar '11

04 Mar '11

hello, is there any way to barge in for things being played by mod_say ? lets say .... you are executing a msgcount macro ... <action application="set" data="playback_terminators=#"/> <action application="phrase" data="msgcount:1234567890"/> when i press # during execution it just suppress 1 prompt than the next one kicks in ... is there any way to do it for entire msgcount macro ?

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checking voicemail without ID?
by Johan Helsingius 04 Mar '11

04 Mar '11

Apologies for what probably is a FAQ, but even with a fair bit of googling I couldn't find a resolution. I am trying to make it possible for users to check their voicemail without knowing their voicemail id (because I want to use a common mailbox for multiple accounts). The standard way seems to be <action application="set" data="voicemail_authorized=${sip_authorized}"/> but it doesn't do the trick for me... Julf

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Fwd: Dialplan in a muddle
by lloyd thomas 04 Mar '11

04 Mar '11

No Joy. Looks like the may be something wrong with the initial companya.tele.co.uk.xml file. Have a look http://pastebin.freeswitch.org/15539 ---------- Forwarded message ---------- From: Garry Matthews <garry.matthews(a)ethermagic.co.uk> Date: 3 March 2011 18:08 Subject: RE: [Freeswitch-users] Dialplan in a muddle To: lloyd thomas <lloydie.t(a)gmail.com> *-* <include> *-* <gateway name="*soho66*"> * * <param name="*username*" value="*1000014877*" /> * * <param name="*realm*" value="*sip.soho66.co.uk:8060*" /> * * <param name="*password*" value="*C9J6U0LO*" /> * * </gateway> * * </include> I’ve made a few changes for security reasons, but it should give you the idea ! If you type >sofia status Then you should see REGED to show that you have registered. Any luck? Garry. *From:* lloyd thomas [mailto:lloydie.t@gmail.com] *Sent:* 03 March 2011 18:02 *To:* Garry Matthews *Subject:* Re: [Freeswitch-users] Dialplan in a muddle Ah Ha, what is your gateway settings (roughly)? On 3 March 2011 17:51, Garry Matthews <garry.matthews(a)ethermagic.co.uk> wrote: I’m not sure if this helps – I just started with Freeswicth and Soho 66 today. I’m a total beginner and have been working from the book. However, I can get calls out using the following file - conf/dialplan/default/01_custom.xml <include> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(0\d{10})$"> <action application="bridge" data="sofia/gateway/soho66/$1"/> </condition> </extension> </include> You then need to - reloadxml This may give you something to compare with. I hope this helps. Garry. *From:* lloyd thomas [mailto:lloydie.t@gmail.com] *Sent:* 03 March 2011 17:04 *To:* freeswitch-users *Subject:* [Freeswitch-users] Dialplan in a muddle Maybe getting a bit ahead of myself but I am trying to set up a multi-company dial plan. Things are working fine internally, but I am not getting the grasp for external calls. The first company xml file is conf/dialplan/companya.tele.co.uk.xml which has the line <X-PRE-PROCESS cmd="include" data="companya.tele.co.uk/*.xml"/> to include xml files in that conf/dialplan/companya.tele.co.uk folder In this folder I have the xml file soho66.co.uk.xml which includes the regex to remove the initial 9 when dialling and extenal number. I cant seem to be able to get a call to go anywhere near connecting to the soho gateway. I think problem is to do with this 2011-03-03 17:00:23.089153 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/200(a)companya.tele.co.uk to enum[ 907973323260(a)companya.tele.co.uk] help me please! ----------------- soho66.co.uk.xml---------------------------- <include> <extension name="Local Dial"> <condition field="destination_number" expression="^9?([2-9]{6})$"> <action application="set" data="effective_caller_id_name=12345678"/> <action application="set" data="effective_caller_id_number=12345678"/> <action application="bridge" data="sofia/gateway/ sip.soho66.co.uk/01869$1(a)sip.soho66.co.uk"/> </condition> </extension> <extension name="Regional 01"> <condition field="destination_number" expression="^9?(01\d{9})$"> <action application="set" data="effective_caller_id_name=12345678"/> <action application="set" data="effective_caller_id_number=12345678"/> <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/> </condition> </extension> <extension name="Regional 02"> <condition field="destination_number" expression="^9?(02\d{9})$$"> <action application="set" data="effective_caller_id_name=12345678"/> <action application="set" data="effective_caller_id_number=123456781"/> <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/> </condition> </extension> <extension name="Mobile"> <condition field="destination_number" expression="^9?(07\d{9})$"> <action application="set" data="effective_caller_id_name=12345678"/> <action application="set" data="effective_caller_id_number=12345678"/> <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/> </condition> </extension> <extension name="International"> <condition field="destination_number" expression="^9?(00\d*))$"> <action application="set" data="effective_caller_id_name=12345678"/> <action application="set" data="effective_caller_id_number=12345678"/> <action application="bridge" data="sofia/gateway/sip.soho66.co.uk/$1"/> </condition> </extension> </include> -----------------eof soho66.co.uk.xml------------------------- -------------conf/sip_profiles/external/companya_soho66.xml ----------------------- <include> <gateway name="sip.soho66.co.uk"> <param name="username" value="12345678"/> <param name="from-user" value="12345678"/> <param name="password" value="ABCDEFGH"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> </include> ------------- eof conf/sip_profiles/external/companya_soho66.xml -----------------------

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FS Webinterfaces - Opinions on wikiPBX, blue.box, fusionPBX
by Christian Benke 04 Mar '11

04 Mar '11

Hi! I'm currently working on a PBX solution for customers of my company(Small to medium businesses) and since i have to make sure my colleagues and customers don't have to dig into the depths of Freeswitch i need a graphical interface for the most mundane tasks. I've spent the last week evaluating FusionPBX and while i was excited at first, i stumbled over a few bugs(Which i could partially solve myself) and had to dig a bit in the code, so i'm not that thrilled anymore. Before i stay with FusionPBX and try to learn some more php and work my way through the code(Is there no ORM in php or did Mark just leave it out?) and add what i'm missing, i'd like to spend another few days trying other projects. Since it takes a while to get to understand a new webinterface and its quirks, i wanted to ask here first if the other options are even worth the time. I think the alternatives are blue.box and WikiPBX? What i'm specifically looking for are: - Proper AAA/groups I want to be able to have a "root"-admin(SuperAdmin in FusionPBX) that has access to the full GUI, a local admin(Admin in FusionPBX) that can change call-relevant features like conferences, dialplans, follow me, voicemail, CDRs and such and finally users, who can only change settings for their own extension and have access to their voicemail and CDRs. Ideally the root-admin should be able to modify what options/features the "lower" groups are able to manipulate(Switch on/off GUI-elements). FusionPBX offers that partially - but the "local admin" has too much rights and i can't change its rights dynamically. - A layer between GUI and configfiles(e.g. a database) I'll have to do some bulk-provisioning with a script from csv-files and if i can write user-information to a DB instead of creating xml-files it would make my life easier - Feature configuration Dialplans are complex and individual and i can comprehend that i won't find an interface that supports all the features i need. FusionPBX was satisfying in this regard, i didn't miss too much and while i couldn't trust a customer to setup his PBX with a complex configuration with Time Conditions, Hunt Groups, Follow-Me etc. i think at least our 2nd level support would be happy to use the gui instead of tinkering with the flatfiles. And i hope my time allows me to implement some of the needed features to FusionPBX(If i don't find a better alternative). Some of them are: Voicemail(Setup and file-access), Huntgroups/FollowMe, Recording, Click2Call, Conferences, CDRs, AutoAttendant/IVR, MOH, TimeConditions, Directory, BossSec/ChefSek and other dubious features asked for by upper mgmt - Phone provisioning It would be nice to be able to add a MAC-adress to a newly created extension and the rest is taken care of by the GUI-scripts. FusionPBX has this feature, but i didn't have time to try it yet(And i'll have to adapt the templates to fit the phones i'm going to use, hope it really works). Nice to have, if not it shouldn't be too much work to write a custom script that can take care of that, given provisioning data from the GUI. - Written in a language i don't have to learn(Python) WikiPBX is based on Django - yes! "http://www.wikipbx.org/ -> Current Status -> CDR issues improved -> 11 Feb 2010" - Crap! Is this project as dead as it seems or is it just so mature there's no further development needed? The screenshots look a bit minimalistic. That's all i need in a perfect world. I guess none of the projects mentioned above can do that for me - but i'd still love to hear your opinions and experiences if it's even worth the time to install/try any of them or stick with FusionPBX! Cheers, Christian

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fs_cli socket connection error
by deniro 04 Mar '11

04 Mar '11

What would be possible reasons for this and how to resolve? running fs 106 on ubuntu 10.04 server /opt/freeswitch/bin/fs_cli -H xxx.xxx.xxx.xxx -P 8021 [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error] was working fine before I installed mod_xml_cdr configure --prefix=/opt/freeswitch --without-libcurl make mod_xml_cdr-install (no errors) in modules.conf.xml <load module="mod_event_socket"/> is there a packaged version of mod_xml_cdr (for fs 1.0.6)? thx deniro--

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weird FS behavior
by envelopes envelopes 04 Mar '11

04 Mar '11

in my FS system, If i only dialed 7 digits, it automatically appended 1347. 347 is not my default area code in vars.xml is FS crazy...??? in the following example, i only dialed : 4562345, but it became 13474562345. 011-03-01 03:22:03.509008 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000(a)192.168.1.115 [fcd969a4-43dc-11e0-8b62-6f20955d792b] 2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000(a)192.168.1.115) Running State Change CS_NEW 2011-03-01 03:22:03.518498 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000(a)192.168.1.115) State NEW 2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4149 Channel sofia/internal/ 1000(a)192.168.1.115 entering state [received][100] 2011-03-01 03:22:03.518498 [DEBUG] sofia.c:4160 Remote SDP: v=0 o=- 28491453 28491453 IN IP4 192.168.1.112 s=- c=IN IP4 192.168.1.112 t=0 0 m=audio 16466 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2011-03-01 03:22:03.524230 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2011-03-01 03:22:03.525384 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2011-03-01 03:22:03.527548 [DEBUG] sofia_glue.c:2354 Set Codec sofia/internal/1000(a)192.168.1.115 PCMU/8000 20 ms 160 samples 2011-03-01 03:22:03.534389 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv payload to 101 2011-03-01 03:22:03.534389 [DEBUG] sofia.c:4306 (sofia/internal/ 1000(a)192.168.1.115) State Change CS_NEW -> CS_INIT 2011-03-01 03:22:03.536796 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/1000(a)192.168.1.115 [BREAK] 2011-03-01 03:22:03.539167 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000(a)192.168.1.115) Running State Change CS_INIT 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000(a)192.168.1.115) State INIT 2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:83 sofia/internal/ 1000(a)192.168.1.115 SOFIA INIT 2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:117 (sofia/internal/ 1000(a)192.168.1.115) State Change CS_INIT -> CS_ROUTING 2011-03-01 03:22:03.540372 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/1000(a)192.168.1.115 [BREAK] 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000(a)192.168.1.115) State INIT going to sleep 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000(a)192.168.1.115) Running State Change CS_ROUTING 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000(a)192.168.1.115) State ROUTING 2011-03-01 03:22:03.540372 [DEBUG] mod_sofia.c:140 sofia/internal/ 1000(a)192.168.1.115 SOFIA ROUTING 2011-03-01 03:22:03.540372 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000(a)192.168.1.115 Standard ROUTING 2011-03-01 03:22:03.540372 [INFO] mod_dialplan_xml.c:418 Processing 1000->13474562345 in context default I did a grep : /etc/freeswitch# find . -name "*.*" | xargs grep 347 ./jingle_profiles/server.xml: <param name="server" value=" jabber.server.org:5347"/> Where is 347 coming from? is it possible of buffer overflow? i am using FS 1.0.6.

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Conference - control actions problem
by Maciej Bylica 03 Mar '11

03 Mar '11

Hello, I am running FreeSWITCH Version 1.0.head (git-06988e1 2011-02-20 15-10-02 -0600) and have some problem with conference module and respective control actions digits. To give you an overview of how it is configured, there is lua script fired in public.xml. The script checks pin code, conf room and at the end run "session:execute("conference", string.format("%s@fsconf", conf_num))". Now coming back to the problem, there are few scenations i would like to cover. 1. When two or more people are in conference and one of them is trying to push # button (RFC2833) then that person is leaving the conference and is hanged up (not released, no BYE). The person who left the conference cannot join again or perform other actions without making the same phone call again. 2011-03-02 11:42:08.863139 [DEBUG] mod_conference.c:2414 Setup timer soft success interval: 20 samples: 160 2011-03-02 11:42:12.283051 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480 2011-03-02 11:42:12.323050 [DEBUG] mod_conference.c:2659 Channel leaving conference, cause: NONE 2. The same as above, but * digit is entered. The person is muted and then unmuted. Next try with * fires muted but this is the point where the problem manifest itself. The procedure of unmuting UA fails. Below you may find relevant logs. The UA is pressing * but without success (no RTP RECV DTMF *). freeswitch@internal> 2011-03-02 11:51:13.423651 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504 2011-03-02 11:51:13.443650 [DEBUG] mod_conference.c:3125 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for play 2011-03-02 11:51:18.163532 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:544 2011-03-02 11:51:18.183531 [DEBUG] mod_conference.c:3125 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-unmuted.wav' for play 2011-03-02 11:51:23.763391 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF *:504 2011-03-02 11:51:23.783390 [DEBUG] mod_conference.c:3125 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for play 3. The same as above but with 0 digit 2011-03-02 12:04:21.064862 [DEBUG] mod_conference.c:2414 Setup timer soft success interval: 20 samples: 160 2011-03-02 12:04:22.863736 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 0:880 2011-03-02 12:04:22.883736 [DEBUG] mod_conference.c:3125 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for play then i cannot get back to the conference 4. The as above but with 6 digit 2011-03-02 12:15:03.244609 [DEBUG] mod_conference.c:2414 Setup timer soft success interval: 20 samples: 160 2011-03-02 12:15:21.602919 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:504 2011-03-02 12:15:22.602893 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:800 2011-03-02 12:15:24.263851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF 6:864 I have disabled all control digits, but the problem still persist. No action is fired but is looks like UA is being muted (micro is off, but i can hear the other person). 2011-03-02 12:24:05.322935 [DEBUG] mod_conference.c:2414 Setup timer soft success interval: 20 samples: 160 2011-03-02 12:24:08.663851 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480 2011-03-02 12:24:11.223787 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:504 2011-03-02 12:24:13.703725 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:480 2011-03-02 12:24:16.043666 [DEBUG] switch_rtp.c:3237 RTP RECV DTMF #:520 Has anybody similar experience or know where the problem may be located. Regards, Maciej

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