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Re: [Freeswitch-users] Freeswitch Build fails (SVN Revision: 10294)
by B Karthik 08 Nov '08

08 Nov '08

Yes it is with make current. i also tried make distclean and rebuild freeswitch from scratch. The same error still persists. B Karthik On Sat, Nov 8, 2008 at 7:11 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > Today's Topics: > > 1. Freeswitch Build fails (SVN Revision: 10294) (B Karthik) > 2. Re: Freeswitch Build fails (SVN Revision: 10294) > (Anthony Minessale) > 3. Re: xml_curl ... (Shelby Ramsey) > 4. Recommendations on ATA's for WinXP and motherboard. > (mszlazak(a)aol.com) > 5. Re: Recommendations on ATA's for WinXP and motherboard. > (Michael Jerris) > > > ---------- Forwarded message ---------- > From: "B Karthik" <carthick84(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sat, 8 Nov 2008 02:03:09 +0530 > Subject: [Freeswitch-users] Freeswitch Build fails (SVN Revision: 10294) > Hi, > > The build fails for the latest svn revision:10294. Error message is as > follows - > > ./.libs/libfreeswitch.so: undefined reference to `switch_dso_open' > ./.libs/libfreeswitch.so: undefined reference to `switch_dso_data_sym' > ./.libs/libfreeswitch.so: undefined reference to `switch_dso_destroy' > > Operating System is Debian Etch. Thanks. > > B Karthik. > > > > ---------- Forwarded message ---------- > From: "Anthony Minessale" <anthony.minessale(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 7 Nov 2008 15:16:04 -0600 > Subject: Re: [Freeswitch-users] Freeswitch Build fails (SVN Revision: > 10294) > is that with make current > > On Fri, Nov 7, 2008 at 2:33 PM, B Karthik <carthick84(a)gmail.com> wrote: > >> Hi, >> >> The build fails for the latest svn revision:10294. Error message is as >> follows - >> >> ./.libs/libfreeswitch.so: undefined reference to `switch_dso_open' >> ./.libs/libfreeswitch.so: undefined reference to `switch_dso_data_sym' >> ./.libs/libfreeswitch.so: undefined reference to `switch_dso_destroy' >> >> Operating System is Debian Etch. Thanks. >> >> B Karthik. >> >> >> _______________________________________________ >> 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 >> >> > > > -- > 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 > > > ---------- Forwarded message ---------- > From: "Shelby Ramsey" <sicfslist(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 7 Nov 2008 15:46:22 -0600 > Subject: Re: [Freeswitch-users] xml_curl ... > Mike, > Thanks for the info on the .cgi ... > > I altered the hangup_after_bridge ... see XML below: > > <?xml version="1.0" encoding="UTF-8" standalone="no"?> > <document type="freeswitch/xml"> > <section name="dialplan" description="FS RESPONSE"> > <context name="public"> > <extension name="17135454263"> > <condition field="destination_number" expression="17135454263"> > <action application="set" data="hangup_after_bridge=false"/> > <action application="set" data="call_timeout=180"/> > <action application="set" data="proxy_media=true"/> > <action application="set" data="pass_rfc2833=true"/> > <action application="set" data="accountcode=SIC_TEST" /> > <action application="set" data="origination_caller_id_name=NULL" > /> > <action application="set" data="origination_caller_id_number=no" > /> > <action application="set" data="effective_caller_id_name=NULL" /> > <action application="set" data="effective_caller_id_number=no" /> > <action application="bridge" data="sofia/internal/ > 7135454263@4.71.122.250:5060" /> > <action application="set" data="origination_caller_id_name=NULL" > /> > <action application="set" > data="origination_caller_id_number=+17133000522" /> > <action application="set" data="effective_caller_id_name=NULL" /> > <action application="set" > data="effective_caller_id_number=+17133000522" /> > <action application="set" data="sip_h_P-Asserted-Identity=<sip: + > 17133000522(a)4.71.122.214;user=phone>" /> > <action application="bridge" data="sofia/internal/ > 7135454263@4.71.122.131:5060" /> > </condition> > </extension> > </context> > </section> > </document> > > I had it set to true based on an example on the wiki (dialplan, example 7): > > <extension name="internal"> > <condition field="destination_number" expression="^1111"> > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" data="sofia/local_profile/1111(a)example1.company.com" /> > <action application="bridge" data="sofia/local_profile/1111(a)example2.company.com" /> > </condition> > </extension> > > > I had previously tried the hangup_after_bridge=false ... > > Any thoughts? > > Thanks for all the help! > > SR > > > On Fri, Nov 7, 2008 at 1:22 PM, Michael Jerris <mike(a)jerris.com> wrote: > >> <action application="set" data="hangup_after_bridge=true"/> >> change to >> >> <action application="set" data="hangup_after_bridge=false"/> >> >> also, if your serving up from xml_curl, you can do the conditions on your >> cgi and just have a blank condition tag, no reason to have the switch do the >> regex as well. >> >> Mike >> >> On Nov 7, 2008, at 1:35 PM, Shelby Ramsey wrote: >> >> Hello, >> I have a question re: xml_curl ... if I reply with this (from the /tmp/ >> file created by fs after xml_curl debug_on): >> >> <?xml version="1.0" encoding="UTF-8" standalone="no"?> >> <document type="freeswitch/xml"> >> <section name="dialplan" description="FS RESPONSE"> >> <context name="public"> >> <extension name="7135454263"> >> <condition field="destination_number" expression="7135454263"> >> <action application="set" data="hangup_after_bridge=true"/> >> <action application="set" data="call_timeout=180"/> >> <action application="set" data="proxy_media=true"/> >> <action application="set" data="pass_rfc2833=true"/> >> <action application="set" data="accountcode=TEST" /> >> <action application="set" data="origination_caller_id_name=NULL" >> /> >> <action application="set" >> data="origination_caller_id_number=7135454263" /> >> <action application="set" data="effective_caller_id_name=NULL" >> /> >> <action application="set" >> data="effective_caller_id_number=7133000522" /> >> <action application="set" data="sip_h_P-Asserted-Identity=<sip: >> +17133000522(a)4.71.122.214;user=phone>" /> >> <action application="bridge" data="sofia/internal/ >> 7135454263(a)4.71.122.250" /> >> <action application="bridge" data="sofia/internal/ >> 7135454263(a)4.71.122.131" /> >> </condition> >> </extension> >> </context> >> </section> >> </document> >> >> FS sends call to sofia/internal/7135454263(a)4.71.122.250 (which returns a >> 404) but does not send to the next "action". >> >> Somewhat of a FS newbie but I thought this should work ... looking at the >> log files I can see this: >> >> 2008-11-07 11:45:00 [DEBUG] switch_core_state_machine.c:140 >> switch_core_standard_on_execute() sofia/external/8885551000(a)4.71.122.250Execute bridge(sofia/internal/ >> 7135454263(a)4.71.122.250) >> >> but it does not do the same thing to the next "action" -- <action >> application="bridge" data="sofia/internal/7135454263(a)4.71.122.131" /> >> >> >> _______________________________________________ >> 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 >> >> > > > ---------- Forwarded message ---------- > From: mszlazak(a)aol.com > To: Freeswitch-users(a)lists.freeswitch.org > Date: Fri, 07 Nov 2008 19:41:41 -0500 > Subject: [Freeswitch-users] Recommendations on ATA's for WinXP and > motherboard. > I would appreciate some recommendations on ATA (analogue telephony > adapters) of either external (e.g. Sipura 3000 which has been updated to a > Linksys box) or internal (Digium 410) type. This would be for a home test > system with one phone line running FS on Windows XP on a motherboard with > shared interrupts (ASUS P5E-VM HDMI). I would like external ATA's over > internal ones but will go the latter if need be. Will the shared interrupts > be a big problem? What are good choices for this set up and approximate > prices? > ------------------------------ > Instant access to the latest & most popular FREE games while you browse > with the Games Toolbar - Download Now!<http://pr.atwola.com/promoclk/100000075x1212904500x1200818240/aol?redir=htt…> > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Fri, 7 Nov 2008 20:41:16 -0500 > Subject: Re: [Freeswitch-users] Recommendations on ATA's for WinXP and > motherboard. > We don't currently have any tdm card support on windows. > > Mike > > On Nov 7, 2008, at 7:41 PM, mszlazak(a)aol.com wrote: > > I would appreciate some recommendations on ATA (analogue telephony >> adapters) of either external (e.g. Sipura 3000 which has been updated to a >> Linksys box) or internal (Digium 410) type. This would be for a home test >> system with one phone line running FS on Windows XP on a motherboard with >> shared interrupts (ASUS P5E-VM HDMI). I would like external ATA's over >> internal ones but will go the latter if need be. Will the shared interrupts >> be a big problem? What are good choices for this set up and approximate >> prices? >> >> > > > _______________________________________________ > 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 > >

1 0

Increase max-sessions and sessions-per-second
by shehzad p 08 Nov '08

08 Nov '08

I need to increase Max Sessions, and sessions-per-second In switch.conf.xml, after increasing "max-session" (originally 1000) to 2000, and then start freeswitch, there it still shows 1000 Max session only?, I can increase it by CLI command "fsctl max_session 2000", but only from configuration file it doesn't work. Same happens for session-per-session I suspect that switch.conf.xml file is not being parsed, whilefreeswitch starts but again all xml files are included in freeswitch.xml file as below: <section name="configuration" description="Various Configuration"> <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/> </section> So what would be the cause. Thanks... MSP -- View this message in context: http://www.nabble.com/Increase-max-sessions-and-sessions-per-second-tp20396… Sent from the Freeswitch-users mailing list archive at Nabble.com.

2 1

[freeswitch-users] answering machine detection
by jonathan augenstine 08 Nov '08

08 Nov '08

I see a description for answer confirmation via DTMF. But, is there any work in progress to implement an actual answering machine detection capability? Jonathan

1 0

Freeswitch Build fails (SVN Revision: 10294)
by B Karthik 07 Nov '08

07 Nov '08

Hi, The build fails for the latest svn revision:10294. Error message is as follows - ./.libs/libfreeswitch.so: undefined reference to `switch_dso_open' ./.libs/libfreeswitch.so: undefined reference to `switch_dso_data_sym' ./.libs/libfreeswitch.so: undefined reference to `switch_dso_destroy' Operating System is Debian Etch. Thanks. B Karthik.

2 1

Recording through Default.xml not working
by Baskar 07 Nov '08

07 Nov '08

Hi, * Recoding is done through default.xml. For past 1 month recoding is working fine. Suddenly for past 3day recording is not working. I did not modify any thing in deafult .xml . * * My deafult.xml file* <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="Local_Extension"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> <action application="answer"/> <action application="info"/> <action application="export" data="Answer-State"/> <action application="log" data="INFO Answer-State [${Answer-State}]\n"/> <action application="record_session" data="$${base_dir}/recordings/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="set" data="answer-state=${Answer-State}"/> <action application="info"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=false"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@ $${domain}"/> <action application="info"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <extension name="Long Distance - wanpipe"> <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="effective_caller_id_number=$1"/> <action application="set" data="dialed_ext=$1"/> <action application="info"/> <action application="set" data="answer-state=${Answer-State}"/> <action application="export" data="effective_caller_id_number=true"/> <action application="export" data="answer-state"/> <action application="set" data="hangup_after_bridge=false"/>  <action application="record_session" data="$${base_dir}/recordings/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bridge" data="sofia/default/$1(a)172.20.191.228"/> </condition> </extension> <extension name="nb_conferences"> <condition field="destination_number" expression="^(30\d{2})$"> <action application="answer"/>  <action application="conference" data="$1@default"/> </condition> </extension> </context> </include> correct me were is wrong -- Warm Regards, N.Baskar

3 4

Different problems with "pizza" demo on newer build.
by mszlazak＠aol.com 06 Nov '08

06 Nov '08

OK, I did my first FS "build" today from download "freeswitch-1.0.1.tar.gz" on Windows XP Pro with Visual C++ 2008 Express instead of using my previous MSI installed build which apparently uses the older FS version 1.0.9570. I'm not getting a crash as I did before with the MSI build on the "pizza" demo. Instead, in the newer build, the demo gets stuck in a loop and can't break out no matter what I say. See below: Any advice? Thank you in advance. Mark. </interpretation> 2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [²²²²½½½½½½½½ε■ε■] 2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:21 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:21 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:25 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:25 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:26 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">TAKEOUT</result> <input>TAKEOUT</input> </interpretation> 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:26 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:27 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:27 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:33 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:33 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:34 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:34 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:34 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:35 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">TAKEOUT</result> <input>TAKEOUT</input> </interpretation> 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:37 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:37 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:42 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:42 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:43 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: DELIVERY, Score: 0 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">DELIVERY</result> <input>DELIVERY</input> </interpretation> 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [DELIVERY] 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:44 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:44 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:50 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:50 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:50 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:51 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">TAKEOUT</result> <input>TAKEOUT</input> </interpretation> 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:22:51 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:51 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:22:51 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:53 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:22:53 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:22:58 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:22:58 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:22:59 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:23:00 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file 2008-11-05 16:23:00 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-05 16:23:01 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: PICKUP, Score: 0 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_order" score="0"> <result name="match">PICKUP</result> <input>PICKUP</input> </interpretation> 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [PICKUP] 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 0/1/78 2008-11-05 16:23:01 [DEBUG] js_modules/SpeechTools.jm:382 console_log() ----We don't understand this 2008-11-05 16:23:03 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2008-11-05 16:23:03 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-05 16:23:04 [DEBUG] sofia.c:197 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/internal/1000(a)10.0.0.2 2008-11-05 16:23:04 [DEBUG] sofia.c:2129 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [calling] 2008-11-05 16:23:04 [DEBUG] sofia.c:2129 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [ready] 2008-11-05 16:23:04 [DEBUG] sofia.c:2133 sofia_handle_sip_i_state() Remote SDP:

3 6

RPID and PAI questions ...
by Shelby Ramsey 06 Nov '08

06 Nov '08

Hello, Couple of quick questions: -- How do I keep FS from sending RPID? And strip the header? And likewise from respecting it? -- How do I add a PAI header -- example here of what I thought would work -- <action application="set" data="sip_h_P-Asserted- Identity=<sip: +17133000522(a)4.71.122.214>" /> ... but it doesn't seem to work (proceeding action is to dial a resource outside of the box -- How do I strip PAI headers from incoming calls? Thanks for any help! Shelby

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More codec playback issues
by Brian Wood 06 Nov '08

06 Nov '08

I'm doing some more testing with various codecs. I'm seeing some more weirdness with the selection of a codec by file extension. I'm testing this on FreeSWITCH trunk 10270. The following script works with the .wav and .gsm extension, but everything else seems to fail. Using lowercase names (such as .g722) fail when recording with a "[ERR] switch_core_file.c:66 switch_core_perform_file_open() Invalid file format [g722] for [/tmp/recording.g722]!". Which is fine, I can use uppercase, but then that plays back garbled audio. This script should beep, record until you stop speaking, beep again, pause, beep, playback, beep, hang up. var allDigits = ""; function on_dtmf(session, type, obj, arg) { try { if (type == "dtmf") { allDigits += obj.digit; console_log("info", "DTMF digit: " + session.name + " [" + obj.digit + "] len [" + obj.duration + "]\n\n"); } } catch (e) { console_log("err", e + "\n"); } return true; } // .gsm works, .wav works //filename = "/tmp/recording.gsm" //filename = "/tmp/recording.wav" // These all result in garbled playback //filename = "/tmp/recording.GSM" filename = "/tmp/recording.G722" /filename = "/tmp/recording.SPEEX" //filename = "/tmp/recording.LPC" //filename = "/tmp/recording.PCMU" //filename = "/tmp/recording.PCMA" //filename = "/tmp/recording.AMR" session.waitForAnswer(10000); session.execute("sleep", "2000"); console_log("info", "*****BEGIN*****"); session.execute("gentones", "%(500,0,800)"); rtn = session.recordFile(filename, on_dtmf, "", 240, 500, 3); session.execute("gentones", "%(500,0,800)"); console_log("info", "*****END*****"); session.execute("sleep", "2000"); session.execute("gentones", "%(500,0,800)"); session.streamFile(filename); session.execute("gentones", "%(500,0,800)"); session.execute("sleep", "1000"); session.hangup();

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Problem in Routing G729A Calls
by shehzad p 06 Nov '08

06 Nov '08

I have to route the inbound calls of G729A codec. Calls comes to my freeswitch with codec G729A (As "annexb=no" is set) But when i route calls to termination gateway, calls are dropped (because of "annexb=no " is not set) Why "annexb=no" is removed while i route the calls? How can I set "annexb=no'? (I am using javascript for routing the calls) Does following SDP variables can help me in solving above problem? How to use those variables? http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation Warm thanks in advance... MSP -- View this message in context: http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.ht… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Inband DTMF Problem
by Klaus Teller 06 Nov '08

06 Nov '08

Hi Folks, It's me again with a DTMF issue. Here is what's going on. I have a remote IVR and i'm writing some code to communicate with it, exercise it, and test it. The remote IVR is playing DTMF as inband. Right now, Freeswitch can detect all DTMF digits from the IVR. The interaction between my code and the IVR is essentially a combination of playing files and sending and reading DTMFs. The issue: whenever my code plays a file or send a DTMF digit, the following attempt to read DTMF fails. By failing i mean that there will be a spurious DTMF detected by Freeswitch that was not sent by the IVR. That is, if i have something like: sendDTMF("3"); getDigits("38373", "#"); // # is the termination character After executing the sendDTMF, getDigits fails because it immediately detects a pound. Yet this pound is not sent by the IVR. I can check this by having wireshark running both on the IVR's computer and on Freeswitch's computer. Any idea what may be causing this problem? Is there some sort of Echo going on or something like that? Thanks, Klaus. -- "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail

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