July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Openzap errors
by Hilda Farrad 30 Jul '08

30 Jul '08

Hello, We are trying to use two Sangoma cards (Analog: A200, E1/T1: A104) with Wanpipe for Freeswitch. We have wanpipe version 3.3.10 and freeswitch 1.0.1 We are running Ubuntu 8.04. We followed the installation and configuration according to the Sangoma wiki as well as the openzap freeswitch wiki without the zaptel installation because during chat we were told we didn't need it, however when we run freeswitch we get these errors: 2008-07-28 16:04:44 [ERR] zap_zt.c:721 zt_init() Cannot open control device 2008-07-28 16:04:44 [ERR] zap_io.c:2134 zap_global_init() Error initilizing zt. Please let me know what can be done to fix this issue.

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Scripting Language
by Ashutosh 30 Jul '08

30 Jul '08

Hi brothers, Now that Freeswitch is there, the only problem remains is of choosing the right language to use with it, as i hold the view, that you can get the best from FS only by using the best and efficient call control language . I am not sure about the following. Please clarify. -For all call control languages (CCL) like js, lua, perl, python, do many calls coming in spawn one instance of the script (a.k.a static class), or each call spawns its own instance of script? -if the answer to above is yes, then is there CCL among lua,perl,python, which has the ability to spawn a single instance, and open and create child threads at will, for all incoming calls ? This is of concern since its essentially bad to have an excellent softswitch, and an inefficent and memory-hogging script working with it. -If there is indeed a CCL to do the above, am i right in saying that the combination of this particular language with FS will yield the highest number of CPS on any given box, (with and without transcoding), compared to other CCLs. Though personally, my choice is of using mod_perl for the above reason because of its "daemonized" nature, but if lua offers the same, i m willing to spend few nights learning lua as well. Please shed light someone. Thanks for all you do! -Ashutosh

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using freeswitch for high volume call traffic
by Ruchir Brahmbhatt 30 Jul '08

30 Jul '08

Hi Everyone, I'm evaluating various solutions for a requirement of softswitch. Main requirements are support for many simultaneous calls, g729 & g723 codecs, CDR, billing, etc. I was wondering if freeswitch is the right solution. I heard that it can handle many simultaneous calls and calls per second is also good. But i have doubts about CDR and billing. Can it do prepaid? Only java script is supported as scripting language? Can it write CDR in mysql table? Can it get configuration from mysql db(like asterisk realtime)? Does it stay in media path when both ends are using g729 or g723 codecs and freeswitch is doing it in passthrough mode?

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Last Call for ClueCon 2008
by Anthony Minessale 29 Jul '08

29 Jul '08

ClueCon 2008 is in one week! Now is the time to register if you have not done so. Please visit http://www.cluecon.com for registration details. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Question on tone_detect
by Michael S Collins 29 Jul '08

29 Jul '08

I'm looking to flesh out the wiki entries on tone_detect and stop_tone_detect and I could use some info. Does stop_tone_detect remove all tone_detects from a session? Does it take any args? Regarding tone_detect (TD) I was wondering if you could feed it a list of freqs and have it do an OR search as opposed the AND search when giving a comma separated list of freqs. Also, could someone tell me the tolerance of the detection? Example: if I detect freq 1100Hz what is the +/- Hz range that will trigger a hit? Lastly, how long must a freq be present to trigger a positive detect? Does it use a scoring model where it counts samples that have the freq present? Thanks, MC Sent from my iPhone

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Re: [Freeswitch-users] Calling web services, JMS or EJB from freeswitch
by Erol Akarsu 29 Jul '08

29 Jul '08

Mike, Can you elaborate some and give some pointers to examples? Where has you used json/rest type calls? Thanks ----- Original Message ---- From: Michael Jerris <mike(a)jerris.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, July 28, 2008 12:00:42 PM Subject: Re: [Freeswitch-users] Calling web services, JMS or EJB from freeswitch I have generally just used json/rest type calls, otherwise you would have to use a lua addin or some code in js. We do have exposed the curl library so all of the http socket handling should be done, just no specifics of whatever protocol you want. Mike On Jul 28, 2008, at 11:57 AM, Brian West wrote: > I think most if not all of this type of functionality is there in both > lua and javascript. I highly recommend lua. > > /b > > On Jul 28, 2008, at 10:52 AM, Erol Akarsu wrote: > >> >> >> I am interested in calling web services,JMS or EJB from inside >> freeswitch, using lua or javascript. >> Is this functionality existing? >> If not, how hard to add this function? >> If it is not so difficult, then I can volunteer. >> >> Best Regards >> >> Erol Akarsu > > Brian West > sip:brian@freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Mod RSS Err for reading news
by unknown 28 Jul '08

28 Jul '08

Hi there, I need your help for the RSS applications under freeswitch. I followed the instructions on http://wiki.freeswitch.org/wiki/Mod_rss and built one extension to test out the RSS in default.xml <extension name="rss"> <condition field="destination_number" expression="^752$"> <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="javascript" data="news.js"/> <action application="rss"/> </condition> </extension> But I still got the errors as follows: 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] switch_core_speech.c:60 switch_core_speech_open() inva lid speech module [cepstral]! 2008-07-28 15:21:09 [ERR] mod_rss.c:266 rss_function() Invalid TTS module! Could anybody direct me to the right track? I have the flite TTS engine and want to use that, no cepstral on the freeswitch. Thanks, Chris

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Contributors notes in source files
by Michael Jerris 28 Jul '08

28 Jul '08

If anyone is a contributor to a source file but their name is not in the list of contributors for that file, please contact me off list so that we can get that corrected. Mike

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Weekly Docs Meeting - Need Items to Discuss
by Michael Collins 28 Jul '08

28 Jul '08

FYI, Here's the link to the weekly documentation conf call page: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_30 We need items to discuss. What's going on out there that needs more or better documentation? What are the hot items? Thanks, MC

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Extension calling.
by Jair Santos 28 Jul '08

28 Jul '08

Hello, I have two users 1000 and 2001. The first is in the same network as FS. The second is in another network. I am trying to : 1000->NAT->internet->NAT->2001 Both users can call PSTN. I can call 2001->1000. When I try to cal 1000->2001 it reachs the configured gateway and I get invalid number. Below are the console outputs. Does anybody know what could be wrong ? thanks Jair Santos 1- Calling 2001->1000 2008-07-28 09:59:29 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)192.168.1.117 [15f1d32c-1f5d-4b8d-a795-825ecd980b95] 2008-07-28 09:59:29 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Cliconnect->2001@default 2008-07-28 09:59:29 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/external/2001 [01c73138-db8f-48a5-8c03-21b15806196e] 2008-07-28 09:59:30 [NOTICE] sofia.c:2491 sofia_handle_sip_i_state() Channel [sofia/external/2001] has been answered 2008-07-28 09:59:30 [NOTICE] switch_ivr_originate.c:1298 switch_ivr_originate() Channel [sofia/internal/1000(a)192.168.1.117] has been answered 2008-07-28 09:59:34 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1000(a)192.168.1.117 [CS_EXECUTE] [NORMAL_CLEARING] 2008-07-28 09:59:34 [NOTICE] switch_ivr_bridge.c:382 audio_bridge_on_exchange_media() Hangup sofia/external/2001 [CS_RESET] [NORMAL_CLEARING] 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 8 (sofia/external/2001) Ended 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/external/2001 [CS_HANGUP] 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 7 (sofia/internal/1000(a)192.168.1.117) Ended 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)192.168.1.117 [CS_HANGUP] 2-Calling 1000->2001 2008-07-28 10:27:20 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->1000@default 2008-07-28 10:27:20 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-07-28 10:27:20 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/2001.2008-07-28-10-27-20.wa v 2008-07-28 10:27:20 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-07-28 10:27:20 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)192.168.1.105;user=phone [5de50898-165d-408b-ba90-ae12c44a6b89] 2008-07-28 10:27:20 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1000(a)192.168.1.105;user=phone! 2008-07-28 10:27:20 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/doublenat/2001@24.67.78.200:5090! 2008-07-28 10:27:20 [NOTICE] switch_ivr_originate.c:1148 switch_ivr_originate() Ring Ready sofia/doublenat/2001@24.67.78.200:5090! 2008-07-28 10:27:23 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/doublenat/2001@24.67.78.200:5090 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-07-28 10:27:23 [NOTICE] switch_ivr_originate.c:1277 switch_ivr_originate() Hangup sofia/internal/1000(a)192.168.1.105;user=phone [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-07-28 10:27:23 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2008-07-28 10:27:23 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 9 (sofia/doublenat/2001@24.67.78.200:5090) Ended 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 10 (sofia/internal/1000(a)192.168.1.105;user=phone) Ended 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)192.168.1.105;user=phone [CS_HANGUP] 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/doublenat/2001@24.67.78.200:5090 [CS_HANGUP] API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED internal2 profile sip:mod_sofia@24.67.78.200:5061 RUNNING (0) external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED doublenat profile sip:mod_sofia@24.67.78.200:5090 RUNNING (0) ============================================================================ =====================

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