December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to run a JS script periodically
by Oscav 03 Dec '09

03 Dec '09

Hi, Someone knows how to run periodically a JS script ?? The purpose is to write to a db some global informations (Global Variables) about FS like every 5 minutes. Thanks. -- View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p266251… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Dialplan behavior
by David Laperle 03 Dec '09

03 Dec '09

Hi guys, i have a weird problem with my dialplans. For the moment, i have only 2 «usable» extensions. They were working #1 yesterday, but this morning i realize i forgot to compile mod_python, so i go back into my source folder and modify the modules.conf to uncomment mod_python, did a make and make install (i did a backup of my conf folder before)! The make and make install worked flawlessly. Then i put back my bkp of conf directory. I restarted the freeswitch service, created my python test dialplan and entered into cli to see what's gonna happen! To my surprise, the call didn't processed to the extension i was dialing. i tried all the other extensions i had, they were all not working!!!! After that i realized that the .xml in freeswitch/dialplan/default/ weren't imported into configuration at startup ... I have read all the documentation about difference between public and default dialplan and i understand them correctly, in public if i include all default folder, it's working again (i can reach all my extensions in default. My extensions are in the correct user_context ... i did nothing since yesterday other than a make && make install after enabling python ... Any other user have an idea why the default/*.xml aren't processed automatically? What could i have done wrong so they are no longer processed? Thanks a lot, David Laperle Administrateur réseau / Network administrator (514) 393-7647 dlaperle(a)rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montréal Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil�gi� et confidentiel est destin� � la seule personne ou entit� � laquelle il est adress�. Pour toute autre personne, toute action prise en rapport � ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'�tes pas la personne autoris�e � recevoir ce courriel, S.V.P. le retourner � l'exp�diteur et le d�truire. Bien que ce courriel ait �t� trait� contre les virus, il est de la responsabilit� du destinataire de s'assurer que l'envoi en est exempt. Nos communications avec vous peuvent contenir des renseignements confidentiels ou prot�g�s par le secret professionnel. Si vous d�sirez que nous communiquions avec vous par un autre moyen de transmission que le courrier �lectronique ordinaire non s�curis�, veuillez nous en aviser. NOTICE: This privileged and confidential email is intended only for the individual or entity to whom it is addressed. With regard to all others, any action related with this email as well as any reading, reproduction, transmission and/or dissemination in whole or in part of the information included in this email is prohibited. If you are not the addressee, immediately return the email to sender prior to destroying all copies. Even if this email is believed to be free from any virus, it is the responsibility of the recipient to make sure that it is virus exempt. Our communications to you may contain confidential information or information protected under solicitor-client privilege. Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you.

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change the remote RTP port after sample rate doesnot match
by Erwin Davis 03 Dec '09

03 Dec '09

Hi, I got a weird issue when I dialed an extension and listen to a recorded voice mail greeting message. After playing a couple of time of the greeting, the FS printed the warning of "sample rate not matching", then send the audio to a different remote RTP port. See the log below, 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@16000hz 1 channels 20ms 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 sofia/internal/1003(a)xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@16000hz 1 channels 20ms 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 sofia/internal/1003(a)xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 sofia/internal/1003(a)xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate doesn't match 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec Activated 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less than minimum record length: 3, discarding it. 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-too-small.wav] (en:en) 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@16000hz 1 channels 20ms 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 sofia/internal/1003(a)xxx.yyy.zzz.31 receive message [ the original codec is wideband 16kHz Speex and the wireshark shows that the FS used the same codec. I used FS 1.04 in fedora 8. I have two questions here, (1) why does FS report "Sample rate doesn't match"? is it a bug or configuration issue? (2) Why does FS change the RTP port ? how to fix it? Thanks, Regards,

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Lua and database access to core_db
by Jon Bruel 03 Dec '09

03 Dec '09

I am trying to rewrite all my javascript scripts into Lua scripts. I have run into the problem of core_db access. This can be achieved with Spidermonkey, but apparently not with Lua. I have tried to get the binary for Lua (using apt-get) but I get an error when I require the sqlite.so: undefined symbol: luaopen_luasql_sqlite, so I'm stuck. So what is a feasible way to manipulate the core database from Lua? I may mention that access to MySQL works perfectly from Lua. Regards Jon

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can't register Inphonex
by John Lalande 03 Dec '09

03 Dec '09

I am new to FS having ditched Asterisk a few weeks ago. I have iptel registered but I cannot get Inphonex to work. I am using the settings from http://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no avail. The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR] sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout [408]." Is there some way to debug this? sofia status displays: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia@192.168.125.15:5080 RUNNING (0) example.com gateway sip:joeuser@example.com NOREG inphonex gateway sip:5285418@sip.inphonex.com FAILED (retry: 28s) iptel gateway sip:jlalande@sip.iptel.org REGED internal profile sip:mod_sofia@192.168.125.15:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) 192.168.125.15 alias internal ALIASED ============================================================================ =====================

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uuid_bridge kills both channels if they are executing java app
by Artem Shiyanov 03 Dec '09

03 Dec '09

Hi there! I've got annoying FS behavior: There are 2 channels executing the same Java application (application itself is an IVR). If I try to bridge them with uuid_bridged then both channels are killed. Here is a log from FS console: uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 (sofia/internal/ 1005(a)192.168.147.130) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1005(a)192.168.147.130 [BREAK] 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 (sofia/internal/ 1001(a)master.agent.starpoundtech.net) State Change CS_EXECUTE -> CS_HIBERNATE 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de freeswitch(a)localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001(a)master.agent.starpoundtec 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1005(a)192.168.147.130 [BREAK] 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal sofia/internal/1001(a)master.agent.starpoundtech.net [BREAK] (FS version is 1.0.4) Any thoughts? Artem

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[local_stream://moh] already broadcasting...broadcast aborted
by Kristian Kielhofner 03 Dec '09

03 Dec '09

Full log and trace here: http://pastebin.freeswitch.org/11062 Pretty standard situation. User calls another user (same profile) and tries to place the call on hold (RFC 3264/sendonly). FS places call on hold and tries to start music but ends with: [local_stream://moh] already broadcasting...broadcast aborted ... and we don't get music. Any ideas why? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Call transfer got broken for me
by Milena 03 Dec '09

03 Dec '09

Hello, It was all ok until yesterday when i updated to svn 15761(last update before that was about 4 days ago), Now I have this issue: someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext 200 200 picks up, then 200 transfers the call to 205 call gets lost (it used to transfer normal until the moment I updated) Today I updated to 15771 and the issue is still there. Can anyone help me figure out what is going on? Call log: http://pastebin.freeswitch.org/11374 thank you

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Gateway issue with no audio
by Henry Huang 03 Dec '09

03 Dec '09

My freeswitch is using public IP. I setup a gateway registering to voipstunt, and put it under internal profile. I tried to make call, and I got no RTP back from the provider... Tried treating NAT issue by changing IP address, internal IP, external IP. But no use, still getting no audio. Finally, I gave up play around with the internal profile and put the gateway *settings under external profile. And magically, it worked.* I am getting audio now. But it leads me to wonders, what's the core difference between external profile and internal profile. Even if I set the external SIP IP and exteranl RTP IP to the public IP in internal profile, I am still getting no audio. Can anyone clear the concept for me here? by the way, I am using freeswitch 1.4 stable version. -- Henry Huang

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Bridging/Connecting Freeswitch servers
by Otis 03 Dec '09

03 Dec '09

Hello I am experimenting with FS and would like to know how to connect two independent servers with user on one beinng able to call users on the other. Do I set each server to be the gateway of the corresponding one ? Pardon me if this has already benn dealt with. My search has drawn a blank Thanks.

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