[asterisk-users] app_conference
Hi list, Has anyone use app_conference? I want to hear what your opinions are. Thnx. attachment: winmail.dat___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Well, if you have control over incoming codecs, yeah sure I recommend it. However, because of the iLBC problem I never solved ( choppy sound ), if you don't have control over codecs joining the conference, may be meet me is still better fo you. Why do you want to move away from meetme? On 9/1/07, Anton Krall [EMAIL PROTECTED] wrote: Hi Moises. So, would you recommend app_conference over meetme? Knowing what you know about it? Saludos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: jueves, 30 de agosto de 2007 09:06 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here: http://osdir.com/ml/telephony.pbx.asterisk.iaxclient.devel/2006-11/msg00016. html But never got response, not sure if is still under development. Right now iLBC voice frames will not work, sound will be choppy, I have not had the time/skills to fix it. Moy On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote: Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Mostly I want to try something new, always testing new stuff.. Ive read some interesting stuff about app_conference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: domingo, 02 de septiembre de 2007 11:15 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Well, if you have control over incoming codecs, yeah sure I recommend it. However, because of the iLBC problem I never solved ( choppy sound ), if you don't have control over codecs joining the conference, may be meet me is still better fo you. Why do you want to move away from meetme? On 9/1/07, Anton Krall [EMAIL PROTECTED] wrote: Hi Moises. So, would you recommend app_conference over meetme? Knowing what you know about it? Saludos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: jueves, 30 de agosto de 2007 09:06 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here: http://osdir.com/ml/telephony.pbx.asterisk.iaxclient.devel/2006-11/msg00016. html But never got response, not sure if is still under development. Right now iLBC voice frames will not work, sound will be choppy, I have not had the time/skills to fix it. Moy On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote: Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Hi Moises. So, would you recommend app_conference over meetme? Knowing what you know about it? Saludos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: jueves, 30 de agosto de 2007 09:06 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_conference Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here: http://osdir.com/ml/telephony.pbx.asterisk.iaxclient.devel/2006-11/msg00016. html But never got response, not sure if is still under development. Right now iLBC voice frames will not work, sound will be choppy, I have not had the time/skills to fix it. Moy On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote: Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference
Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here: http://osdir.com/ml/telephony.pbx.asterisk.iaxclient.devel/2006-11/msg00016.html But never got response, not sure if is still under development. Right now iLBC voice frames will not work, sound will be choppy, I have not had the time/skills to fix it. Moy On 8/30/07, Anton Krall [EMAIL PROTECTED] wrote: Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference and asterisk 1.2.24
Is app_conference designed only for 1.4? I tried compiling against 1.2.24 and but get a no such file while looking for autoconf.h which is a file only used in 1.4... anybody running app_conference on 1.2? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference not working for me
I'm having trouble getting app_conference to work and I'm feeling pretty clueless right know. With no flags, it doesn't exit when I press '#.' With flags passed as d, it just ignores '#.' With flags passed as MTV, it crashes Asterisk when I press '#.' Any clues would be appreciated :) Here's how I'm invoking conference: exten = *,n,conference(test) or exten = *,n,conference(test|d) or exten = *,n,conference(test|MTV) Here's the console output with no flags: -- Accepting AUTHENTICATED call from a.b.c.d: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw), priority = mine -- Executing Conference(IAX2/a.b.c.d:1030-4, test) in new stack Sep 4 12:53:41 ERROR[10454]: frame.c:386 convert_frame: unable to translate frame Here's what gets syslogged: Sep 4 12:53:35 dt-ext asterisk[10222]: VERBOSE[10227]: -- Accepting AUTHENTICATED call from a.b.c.d: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw), priority = mine Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10225]: chan_iax2.c:9434 in iax2_devicestate: Checking device state for device a.b.c.d Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10225]: devicestate.c:187 in do_state_change: Changing state for IAX2/a.b.c.d:1030 - state 4 (Invalid) Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: pbx.c:1677 in pbx_extension_helper: Launching 'Conference' Sep 4 12:53:35 dt-ext asterisk[10222]: VERBOSE[10453]: -- Executing Conference(IAX2/a.b.c.d:1030-4, test) in new stack Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:415 in member_exec: [ $Revision: 1.9 $ ] begin processing member thread, channel = IAX2/a.b.c.d:1030-4 Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: chan_iax2.c:3370 in iax2_answer: Answering IAX2 call Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:742 in create_member: attempting to parse passed params, stringp = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:793 in create_member: parsed data params, id = test, flags = , priority = 0, vad_prob_start = 0.05, vad_prob_continue = 0.02 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:1077 in create_member: created member, type = S, priority = 0, readformat = 4 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:451 in member_exec: CHANNEL INFO, CHANNEL = IAX2/a.b.c.d:1030-4, DNID = *, CALLER_ID = 21012006, ANI = 21012006 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:454 in member_exec: CHANNEL CODECS, CHANNEL = IAX2/a.b.c.d:1030-4, NATIVE = 4, READ = 4, WRITE = 4 Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: channel.c:2376 in set_format: Set channel IAX2/a.b.c.d:1030-4 to read format ulaw Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: channel.c:2376 in set_format: Set channel IAX2/a.b.c.d:1030-4 to write format ulaw Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:504 in start_conference: attempting to find requested conference Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:548 in find_conf: conflist has not yet been initialized, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:511 in start_conference: attempting to create requested conference Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:583 in create_conf: entered create_conf, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to unknown Sep 4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to alaw Sep 4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to unknown Sep 4 12:53:35 dt-ext last message repeated 5 times Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:796 in add_member: member added to conference, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:646 in create_conf: added new conference to conflist, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:663 in create_conf: started conference thread for conference, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:514 in member_exec: begin member event loop, channel = IAX2/a.b.c.d:1030-4 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:532 in member_exec: Conference Members: 1 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:538 in member_exec: Quiet debug 0 - 0 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:546 in member_exec: skipping entry message on IAX2/a.b.c.d:1030-4 Sep 4
Re: [asterisk-users] app_conference not working for me
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: I'm having trouble getting app_conference to work and I'm feeling pretty clueless right know. Probably the iaxclient list would be the better forum to discuss this as its not in the Asterisk codebase. To sign up for the iaxclient mailing list go to: https://lists.sourceforge.net/lists/listinfo/iaxclient-devel - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE/J6cS6d5vy0jeVcRArIzAJ9GxywjnYuC8k/bOOFsqDqaE6VF/wCbBD10 oXJfOkjFL/MUxpbz+4bDVNE= =W+n6 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
In this sections there are a context [conferences] Where Can I put this lines? in extension.conf?2006/8/19, RR [EMAIL PROTECTED] :Follow the instructions here: http://www.voip-info.org/wiki/view/Asterisk+app_conferenceThere's no config file where conferences are stored. You need to addthem to astdb using the 'database' CLI command like so: database putconferences 1234 9 Look at the setting up conferences section in the Wiki___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference
HelloI installed asterisk with app_conferenceBut How and Where Can I set an conference?Thanks for your answers!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference
Follow the instructions here: http://www.voip-info.org/wiki/view/Asterisk+app_conference There's no config file where conferences are stored. You need to add them to astdb using the 'database' CLI command like so: database put conferences 1234 9 Look at the setting up conferences section in the Wiki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident.are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? j- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
[EMAIL PROTECTED] wrote: On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident. are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
It really depends on the application. app_conference does wonderfully for long conferences without a lot of entry/exit and no playing of audio files. The issues with the double-free crashes that we've had all seem to be caused by playing of audio files(like the entry/exit sounds or the DTMF broadcast). But these functions use the app_conference code that was already existing to play audio files from manager API commands so the issue was there it just wasn't as tested because not many people use the manager command a lot to play audio in conferences. The other issue we had with app_conference was using it in high-volume VICIDIAL outbound production(thousands of entry/exit actions per hour) where it would always fail after 1-8 hours. In this case there wasn't a crash, but strangely app_conference just seemed to stop working like the engine died. Everything else in Asterisk kept working but you couldn't do anything in app_conference without stopping and starting Asterisk again. MATT--- On 7/12/06, jeff oconnell [EMAIL PROTECTED] wrote: On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident. are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? j- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
interesting. i didn't realize the problem seems to specifically be the sound playback via the manager interface.a couple weeks ago, my asterisk on my dev box crashed, i did some preliminary investigation, but since we hadn't had any problems in production or qa, i chalked it up to me messing up my dev install somehow. anyway, here's the backtrace of the core:(gdb) bt#0 0x00df71bb in ?? () from /lib/libgcc_s.so.1#1 0x080625ad in ast_deactivate_generator (chan=0x815fb40) at channel.c:1382#2 0x0806d77a in ast_openstream_full (chan=0x815fb40, filename=0x815dcec tones-exit, preflang=0x0, asis=0) at file.c:494#3 0x0806d835 in ast_openstream (chan=0x71bfb8a6, filename=0x71bfb8a6 Address0x71bfb8a6 out of bounds, preflang=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at file.c:467#4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c)at cli.c:225#5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 outof bounds) at cli.c:1364#6 0x080b2f8a in action_command (s=0x815c448, m=0xb7b97420) at manager.c:927#7 0x080b7b81 in process_message (s=0x815c448, m=0xb7b97420) at manager.c:1305#8 0x080b83cf in session_do (data="" at manager.c:1401#9 0x00b44341 in start_thread () from /lib/tls/libpthread.so.0#10 0x009096fe in clone () from /lib/tls/libc.so.6the interesting lines to me are #4 and #5:#4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c) at cli.c:225 #5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at cli.c:1364 line 4 becauset the argc passed into conference_play_sound() is so large, and line 5 because there seems to be an out--of-bounds problem in the asterisk code ( i.e. before the app_conference code is called ). based on what you said in your last post, i'm going to look at this more.if you have any thoughts on my backtrace/analysis, let me know.j-On 7/12/06, Matt Florell [EMAIL PROTECTED] wrote: The issues with the double-free crashes that we've had all seem to becaused by playing of audio files(like the entry/exit sounds or theDTMF broadcast). But these functions use the app_conference code thatwas already existing to play audio files from manager API commands so the issue was there it just wasn't as tested because not many peopleuse the manager command a lot to play audio in conferences. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
Hello, My backtraces never actually mention play_sound, but the crashes only happen right after app_conference attempts to play out DTMF tines with the playing function. Here's the backtrace for two of the crashes that we had with app_conference: http://205.201.151.24/files/app_conference-crash-2006-06-02.txt http://205.201.151.24/files/app_conference-crash-2006-06-05.txt MATT--- On 7/12/06, jeff oconnell [EMAIL PROTECTED] wrote: interesting. i didn't realize the problem seems to specifically be the sound playback via the manager interface. a couple weeks ago, my asterisk on my dev box crashed, i did some preliminary investigation, but since we hadn't had any problems in production or qa, i chalked it up to me messing up my dev install somehow. anyway, here's the backtrace of the core: (gdb) bt #0 0x00df71bb in ?? () from /lib/libgcc_s.so.1 #1 0x080625ad in ast_deactivate_generator (chan=0x815fb40) at channel.c:1382 #2 0x0806d77a in ast_openstream_full (chan=0x815fb40, filename=0x815dcec tones-exit, preflang=0x0, asis=0) at file.c:494 #3 0x0806d835 in ast_openstream (chan=0x71bfb8a6, filename=0x71bfb8a6 Address 0x71bfb8a6 out of bounds, preflang=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at file.c:467 #4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c) at cli.c:225 #5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at cli.c:1364 #6 0x080b2f8a in action_command (s=0x815c448, m=0xb7b97420) at manager.c:927 #7 0x080b7b81 in process_message (s=0x815c448, m=0xb7b97420) at manager.c:1305 #8 0x080b83cf in session_do (data=0x815c448) at manager.c:1401 #9 0x00b44341 in start_thread () from /lib/tls/libpthread.so.0 #10 0x009096fe in clone () from /lib/tls/libc.so.6 the interesting lines to me are #4 and #5: #4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c) at cli.c:225 #5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at cli.c:1364 line 4 becauset the argc passed into conference_play_sound() is so large, and line 5 because there seems to be an out--of-bounds problem in the asterisk code ( i.e. before the app_conference code is called ). based on what you said in your last post, i'm going to look at this more. if you have any thoughts on my backtrace/analysis, let me know. j- On 7/12/06, Matt Florell [EMAIL PROTECTED] wrote: The issues with the double-free crashes that we've had all seem to be caused by playing of audio files(like the entry/exit sounds or the DTMF broadcast). But these functions use the app_conference code that was already existing to play audio files from manager API commands so the issue was there it just wasn't as tested because not many people use the manager command a lot to play audio in conferences. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
Matt Florell [EMAIL PROTECTED] wrote: My backtraces never actually mention play_sound, but the crashes only happen right after app_conference attempts to play out DTMF tines with the playing function. This is because Malloc isn't crashing when the mistake is made. It crashes later because of the out of bounds write or double free has corrupted its memory structures. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
Henry J. Cobb wrote: I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. I hate to me-too, but my experience was identical. Crash after crash, and I tried everything that was suggested (limiting codecs, primarily). Something is weird there in that for some it appears to work perfectly, for others not at all. . . FWIW. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
thanks brian, this is all really helpful feedback!just to be clear, which app_conference code were you using?the svn trunk version from sourceforge? or the VD_app_conference matt's been working on? j-On 7/12/06, Brian Capouch [EMAIL PROTECTED] wrote: I hate to me-too, but my experience was identical.Crash after crash,and I tried everything that was suggested (limiting codecs, primarily).Something is weird there in that for some it appears to work perfectly, for others not at all. . . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
Brian Capouch [EMAIL PROTECTED] wrote: Henry J. Cobb wrote: I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. I hate to me-too, but my experience was identical. Crash after crash, and I tried everything that was suggested (limiting codecs, primarily). Something is weird there in that for some it appears to work perfectly, for others not at all. . . This is starting to stink a lot more like a memory overrun error than a double free error. This looks like a job for Electric Fence. http://www.die.net/doc/linux/man/man3/efence.3.html -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
yeah, if you have the source code around, look for a file called 'VICIDIAL.txt' in the app_conference directory.j-On 7/12/06, Brian Capouch [EMAIL PROTECTED] wrote:jeff oconnell wrote: thanks brian, this is all really helpful feedback! just to be clear, which app_conference code were you using? the svn trunk version from sourceforge? or the VD_app_conference matt's been working on? Yikes.I don't remember.Do you know if I can tell by looking somewhere in the source code?Thx.B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
that means you've been using matt's modified version. you can get the latest stable version ( minus matt's new dtmf features, etc. ) from the sourceforge subversion repository: svn co https://svn.sourceforge.net/svnroot/iaxclient/trunk/app_conference give it a whirl and let us know if it works for you. j- On 7/12/06, Brian Capouch [EMAIL PROTECTED] wrote: jeff oconnell wrote: yeah, if you have the source code around, look for a file called 'VICIDIAL.txt' in the app_conference directory.Affirmative.Which does that indicate that I've built? B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
henry, did you have any luck setting this up? i'm actually working right now to _suppress_ dtmf clicks in app_conference, and would be happy to look at the dtmf pass-through, if you're still in need. j- On 5/29/06, Henry J. Cobb [EMAIL PROTECTED] wrote: We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference DTMFs?
I have written such a modification into app_conference. It allows the option of rebroadcasting DTMF tones and/or RFC frames to participants if enabled. There are also a few other modifications in the version that I am using. I released it a month ago and have used it on a few servers since. It works well but app_conference has some memory issues that cause it to crash a couple times a week(I have sent gdb backtraces to the iaxdev list and will send to anyone else that's goo with debugging) http://sourceforge.net/project/shownotes.php?release_id=421962 Let me know what you think, MATT--- On 5/29/06, Henry J. Cobb [EMAIL PROTECTED] wrote: We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
matt, i was looking at your dtmf changes today. they look pretty interesting. right now i'm working on a scheme for cleaning up the clicking we hear when dtmf tones are not fully filtered by front-end asterisk servers. meetme seems to do this by calling: ast_channel_setoption( chan, AST_OPTION_TONE_VERIFY, [...] ) but this doesn't work for us because of the way our infrastructure is set up. anyway, from what i understand, mihai is going to look at your changes more, but while i'm in the code, i'll also take a look and see if i can figure out what your memory issues are... j- On 7/11/06, Matt Florell [EMAIL PROTECTED] wrote: I have written such a modification into app_conference. It allows the option of rebroadcasting DTMF tones and/or RFC frames to participants if enabled. There are also a few other modifications in the version that I am using. I released it a month ago and have used it on a few servers since. It works well but app_conference has some memory issues that cause it to crash a couple times a week(I have sent gdb backtraces to the iaxdev list and will send to anyone else that's goo with debugging) http://sourceforge.net/project/shownotes.php?release_id=421962 Let me know what you think, MATT--- On 5/29/06, Henry J. Cobb [EMAIL PROTECTED] wrote: We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
Sounds good, let me know if you want the gdb bt full output from the core dumps that I have. The DTMF broadcasting was a workaround to be able to use a non-Zap channel in a conference(non-Zap channels in a meetme cannot always send DTMF and it's strange design made it very difficult to alter). The ability to broadcast both/either RFC and/or inband is a result of having to be able to work with Zap and VOIP channels in the same conference. I also changed the launching of app_conference in the extensions.conf to be more like meetme and added the option for entry/exit sounds so that it could be more of a drop-in replacement for meetme. The VD_app_conference package is tested enough to be a fully functional replacement for conferencing in the VICIDIAL call center app which is the project I did all of these changes for. Let me know if you have any questions about it, MATT--- On 7/11/06, jeff oconnell [EMAIL PROTECTED] wrote: matt, i was looking at your dtmf changes today. they look pretty interesting. right now i'm working on a scheme for cleaning up the clicking we hear when dtmf tones are not fully filtered by front-end asterisk servers. meetme seems to do this by calling: ast_channel_setoption( chan, AST_OPTION_TONE_VERIFY, [...] ) but this doesn't work for us because of the way our infrastructure is set up. anyway, from what i understand, mihai is going to look at your changes more, but while i'm in the code, i'll also take a look and see if i can figure out what your memory issues are... j- On 7/11/06, Matt Florell [EMAIL PROTECTED] wrote: I have written such a modification into app_conference. It allows the option of rebroadcasting DTMF tones and/or RFC frames to participants if enabled. There are also a few other modifications in the version that I am using. I released it a month ago and have used it on a few servers since. It works well but app_conference has some memory issues that cause it to crash a couple times a week(I have sent gdb backtraces to the iaxdev list and will send to anyone else that's goo with debugging) http://sourceforge.net/project/shownotes.php?release_id=421962 Let me know what you think, MATT--- On 5/29/06, Henry J. Cobb [EMAIL PROTECTED] wrote: We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
jeff oconnell [EMAIL PROTECTED] wrote: but while i'm in the code, i'll also take a look and see if i can figure out what your memory issues are... When I've tried it, app_conference always crashed within the hour. I think that the entire Asterisk server, including app_conference, needs to be compiled with one of the debugging malloc libraries because it might be that you are returning something to Asterisk in such a way that it does either a double free or a free of non-mallocated memory. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference sources?
The CVS server for app_conference seems to be down. Can somebody mail me a recent copy of the sources please? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference DTMFs?
We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference compiling for asterisk
Hi all, today I download the app_conference from iaxclient-dvs. I edit the Makefile to my paths: INSTALL_PREFIX := /usr INSTALL_MODULES_DIR := $(INSTALL_PREFIX)/lib/asterisk/modules ASTERISK_INCLUDE_DIR := $(INSTALL_PREFIX)/src/asterisk-1.2.0-rc2/ include/asterisk and then try make, but I only get the following errors: [EMAIL PROTECTED] app_conference]# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/ usr/src/asterisk-1.2.0-rc2/include/asterisk -D_REENTRANT - D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop- arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse, 387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c In file included from /usr/include/unistd.h:26, from /usr/include/asterisk/channel.h:89, from /usr/include/asterisk/pbx.h:27, from app_conference.h:23, from app_conference.c:19: /usr/src/asterisk-1.2.0-rc2/include/asterisk/features.h:44: Fehler: syntax error before »AST_LIST_ENTRY« In file included from /usr/include/bits/types.h:31, from /usr/include/unistd.h:186, from /usr/include/asterisk/channel.h:89, from /usr/include/asterisk/pbx.h:27, from app_conference.h:23, from app_conference.c:19: /usr/lib/gcc/i386-redhat-linux/4.0.1/include/stddef.h:214: Fehler: syntax error before »typedef« In file included from /usr/include/asterisk/channel.h:89, from /usr/include/asterisk/pbx.h:27, from app_conference.h:23, from app_conference.c:19: /usr/include/unistd.h:256: Fehler: syntax error before »__THROW« /usr/include/unistd.h:287: Fehler: syntax error before »__THROW« /usr/include/unistd.h:313: Fehler: syntax error before »__wur« /usr/include/unistd.h:319: Fehler: syntax error before »__wur« /usr/include/unistd.h:370: Fehler: syntax error before »__THROW« /usr/include/unistd.h:379: Fehler: syntax error before »__THROW« /usr/include/unistd.h:420: Fehler: syntax error before »__THROW« /usr/include/unistd.h:435: Fehler: syntax error before »__THROW« /usr/include/unistd.h:449: Fehler: syntax error before »__THROW« /usr/include/unistd.h:468: Fehler: syntax error before »__THROW« /usr/include/unistd.h:471: Fehler: syntax error before »__THROW« /usr/include/unistd.h:483: Fehler: syntax error before »__THROW« /usr/include/unistd.h:495: Fehler: syntax error before »__THROW« /usr/include/unistd.h:500: Fehler: syntax error before »__THROW« /usr/include/unistd.h:505: Fehler: syntax error before »__THROW« /usr/include/unistd.h:510: Fehler: syntax error before »__THROW« /usr/include/unistd.h:516: Fehler: syntax error before »__THROW« In file included from /usr/include/asterisk/channel.h:89, from /usr/include/asterisk/pbx.h:27, from app_conference.h:23, from app_conference.c:19: /usr/include/unistd.h:536: Fehler: syntax error before »__THROW« /usr/include/unistd.h:539: Fehler: syntax error before »__THROW« /usr/include/unistd.h:542: Fehler: syntax error before »__THROW« /usr/include/unistd.h:551: Fehler: syntax error before »__THROW« /usr/include/unistd.h:554: Fehler: syntax error before »__THROW« /usr/include/unistd.h:559: Fehler: syntax error before »__THROW« /usr/include/unistd.h:569: Fehler: syntax error before »__THROW« /usr/include/unistd.h:578: Fehler: syntax error before »__THROW« /usr/include/unistd.h:612: Fehler: syntax error before »__THROW« /usr/include/unistd.h:620: Fehler: syntax error before »__THROW« /usr/include/unistd.h:623: Fehler: syntax error before »__THROW« Can anybody help? I tried different options, but I dont find the mistake Best regards and many thanks Dominik Simon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference and AGI
At 15:21 06/07/2005 +0200, Tobias Wolf wrote:
Hi,
i was successful in compiling app_conference and setting up an conference
was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from inside
the conference. So, if i dialed into an conference i want to be able to
press '*' and then the actual discussion is muted for me and i and menu is
read to me. Something like the ${MEETME_AGI_BACKGROUND} in MeetMe.
Thx in advance :)
Tobias Wolf
Looking at apps/app_meetme.c, I saw that there is a POUNDEXIT option
that when set will kick a user when she hits #, you use 'p' as an option
when invoking Meetme in the dial plan.
There is another option, STARMENU, that enables an admin menu when user
hits * ('s' option)
I guess that you could either change your mind and use # or patch
app_meetme to accept both # and * (when STARMENU is not enable) or
patch app_meetme to inverse the roles of # and *. Ideally you want both
DTMFs to be configurable instead of hard coded, but that's another story.
Once you get what you want there, i.e. the ability to leave the conference,
you will handle the IVR in the dialplan I suppose. When done, you get back
to the conference room in meetme (assuming you tracked it).
But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
that unfortunately it does not work for SIP channels due to the mixing
not being done in the zaptel driver but app_meetme itself, sort of, AFAIK).
Hope this helps,
Yours,
JeanHuguesRobert
-
Web: http://hdl.handle.net/1030.37/1.1
Phone: +33 (0) 4 92 27 74 17
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[Asterisk-Users] app_conference and AGI
Hi,
i was successful in compiling app_conference and setting up an
conference was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from
inside the conference. So, if i dialed into an conference i want to be
able to press '*' and then the actual discussion is muted for me and i
and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in
MeetMe.
Thx in advance :)
Tobias Wolf
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[Asterisk-Users] app_conference, CVS HEAD, SIP and Xen
I have Asterisk running in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load the module into Asterisk and even connect to a conference channel with two phones. But each phone cannot send audio to each other. They just connect and go silent. Is this the current state of app_conferences development? I have read few comments online that sound like some people are using app_conference with sound. -- Lee Azzarello Network Engineer Progressive Solutions +1 212 937 8939 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference, CVS HEAD, SIP and Xen
Lee Azzarello wrote: I have Asterisk running in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load the module into Asterisk and even connect to a conference channel with two phones. But each phone cannot send audio to each other. They just connect and go silent. Is this the current state of app_conferences development? I have read few comments online that sound like some people are using app_conference with sound. No, app_conference should work fine. I can't say what's happening in your case, though.. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_conference in dial plan?
Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferencing... ; Make as many of these contexts as you have seperate conference bridges ; change conferencename in each [conf-conferencename] exten = join,1,System(/opt/asterisk/bin/conference-announce conferencename in) exten = join,2,Conference(conferencename/S/1) exten = h,1,System(/opt/asterisk/bin/conference-announce conferencename out) [confhelper] ; make one of these extensions per seperate conference bridge exten = conf-conferencename,1,Conference(conferencename/S/1) exten = in,1,Answer() ; if I use Playback here instead of BackGround, asterisk crashes exten = in,2,BackGround(conf-announce) exten = in,3,ResponseTimeout(5) exten = in,4,Hangup() exten = out,1,Answer() exten = out,2,BackGround(conf-leave) exten = out,3,ResponseTimeout(5) exten = out,4,Hangup() how do I setup up app_conference to respond to an extention? Just a real simple example to get me started would be appreciated... I've tried a few things along the lines of the example meetme extention ie exten = 901,1,app_conference(901||1234) or exten = 901,1,cmd_conference(901||1234) But I guess its expecting too much to think that this would fireup app_conference Thanks in advance for any help. Cheers, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App_conference in dial plan?
exten = 901,1,Conference(Internal Test Conference/S/1) Looks like it does the job... Mark Benson wrote: Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferencing... ; Make as many of these contexts as you have seperate conference bridges ; change conferencename in each [conf-conferencename] exten = join,1,System(/opt/asterisk/bin/conference-announce conferencename in) exten = join,2,Conference(conferencename/S/1) exten = h,1,System(/opt/asterisk/bin/conference-announce conferencename out) [confhelper] ; make one of these extensions per seperate conference bridge exten = conf-conferencename,1,Conference(conferencename/S/1) exten = in,1,Answer() ; if I use Playback here instead of BackGround, asterisk crashes exten = in,2,BackGround(conf-announce) exten = in,3,ResponseTimeout(5) exten = in,4,Hangup() exten = out,1,Answer() exten = out,2,BackGround(conf-leave) exten = out,3,ResponseTimeout(5) exten = out,4,Hangup() how do I setup up app_conference to respond to an extention? Just a real simple example to get me started would be appreciated... I've tried a few things along the lines of the example meetme extention ie exten = 901,1,app_conference(901||1234) or exten = 901,1,cmd_conference(901||1234) But I guess its expecting too much to think that this would fireup app_conference Thanks in advance for any help. Cheers, Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_Conference
Anyone tried to build app_conference lately? I'm trying to setup a large conference where i speaker can talk to many listeners, for example 1 speaker and about 100 listeners (who can not speak back to the speaker, 1 way audio only) However, if i try to build app_conference against 1.0.6 or 1.0.7 it won't compile with an error message: make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o app_conference.o app_conference.c gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o conference.o conference.c conference.c: In function `create_conf': conference.c:614: warning: implicit declaration of function `__use_ast_pthread_create_instead__' gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o member.o member.c member.c: In function `member_exec': member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:165: warning: unused variable `ignore_speex_count' make: *** [member.o] Error 1 _ Direct antwoord op je vragen: gebruik MSN Messenger http://messenger.msn.nl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App_Conference
I believe you need to modify a little bit member.c file in CVS version they use cid, but in stable version callerid. Just replace properly cid with callerid. It should help with that problem. For example: chan-cid.cid_num change to chan-callerid On Mon, 2005-04-18 at 10:04, E rikje wrote: Anyone tried to build app_conference lately? I'm trying to setup a large conference where i speaker can talk to many listeners, for example 1 speaker and about 100 listeners (who can not speak back to the speaker, 1 way audio only) However, if i try to build app_conference against 1.0.6 or 1.0.7 it won't compile with an error message: make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o app_conference.o app_conference.c gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o conference.o conference.c conference.c: In function `create_conf': conference.c:614: warning: implicit declaration of function `__use_ast_pthread_create_instead__' gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o member.o member.c member.c: In function `member_exec': member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:165: warning: unused variable `ignore_speex_count' make: *** [member.o] Error 1 _ Direct antwoord op je vragen: gebruik MSN Messenger http://messenger.msn.nl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference compile?
Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference
I tried to get app_conference running tonight, but it seems to crash with segmentation faults, every time the second user enters the system. Here is the console output (ip addresses removed) from the session, including gdb output at the segmentation fault: -- Accepting unauthenticated call from XXX.XXX.XXX.XXX, requested format = 2, actual format = 1024 Nov 5 04:35:35 NOTICE[180236]: channel.c:284 ast_alloc_uniqueid: uid = asterisk-11277-1099629335.1 [New Thread 458781 (LWP 11313)] -- Executing Conference(IAX2/[EMAIL PROTECTED]/1, Test/M/0) in new stack [New Thread 475166 (LWP 11314)] -- Accepting unauthenticated call from XXX.XXX.XXX.XXX, requested format = 2, actual format = 1024 Nov 5 04:35:37 NOTICE[180236]: channel.c:284 ast_alloc_uniqueid: uid = asterisk-11277-1099629337.2 [New Thread 491551 (LWP 11315)] Nov 5 04:35:37 NOTICE[458781]: chan_iax2.c:2473 iax2_read: I should never be called! -- Executing Conference(IAX2/[EMAIL PROTECTED]/5, Test/M/0) in new stack Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 475166 (LWP 11314)] 0x40189cd5 in mallopt () from /lib/libc.so.6 This is Debian Sarge, libc6 2.3.2.ds1-18 asterisk 1.0.2 app_conference from cvs tonight (though i had to fix the mutex stuff etc. to get it compiled). Any suggestions ? Kind regards, Martin List-Petersen -- woot Put *that* in you .sig and smoke it, Knghtbrd. Culus You know he will read this : woot heheheheh. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference
Thanks to all who have helped me build and test out Asterisk installation thus far. I needed to move my * installation to a new box , due to the fact my test machine would not support PCI 2.2 ( which I am told is required to use my TDM11B). I have * up and running and I am attempting to compile the app_conference source. The MeetMe app has too much echo. I am running Debian 2.4.26 and get tons of compile errors. If I compile right from the CVS of app_conference I get: chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: error: invalid option `abi=altivec' cc1: error: invalid option `dynamic-no-pic' cc1: error: unrecognized option `-faltivec' cc1: error: bad value (7450) for -mcpu= switch cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) make: *** [app_conference.o] Error 1 I then fix the mcpu ( I am on a Pentium4 Box). I comment out the line. I run make clean and make and then get the following. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [conference.o] Error 1 I change my two lines in conference.c as per http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html I run make clean, make and get the following error chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o member.o member.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) member.c: In function `member_exec': member.c:76: error: structure has no member named `dnid' member.c:76: error: structure has no member named `callerid' member.c:76: error: structure has no member named `ani' make: *** [member.o] Error 1 I have edited my member.c to remove any reference to the dnid,callerid and ani , and it compiles. But when someone connects to the conference * crashes. I know I am missing something simple ( I hope) I have also followed the instructions on http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html to no avail. I have also copied my app_conference files from another Asterisk box ( it compiles fine on that box). On my new box it will not compile. TIA Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference
Shawn Dillon wrote: Thanks to all who have helped me build and test out Asterisk installation thus far. I needed to move my * installation to a new box , due to the fact my test machine would not support PCI 2.2 ( which I am told is required to use my TDM11B). I have * up and running and I am attempting to compile the app_conference source. The MeetMe app has too much echo. I am running Debian 2.4.26 and get tons of compile errors. If I compile right from the CVS of app_conference I get: chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: error: invalid option `abi=altivec' cc1: error: invalid option `dynamic-no-pic' cc1: error: unrecognized option `-faltivec' cc1: error: bad value (7450) for -mcpu= switch cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) make: *** [app_conference.o] Error 1 I then fix the mcpu ( I am on a Pentium4 Box). I comment out the line. I run make clean and make and then get the following. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [conference.o] Error 1 I change my two lines in conference.c as per http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html I run make clean, make and get the following error chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o member.o member.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) member.c: In function `member_exec': member.c:76: error: structure has no member named `dnid' member.c:76: error: structure has no member named `callerid' member.c:76: error: structure has no member named `ani' make: *** [member.o] Error 1 I have edited my member.c to remove any reference to the dnid,callerid and ani , and it compiles. But when someone connects to the conference * crashes. Did you remove the whole ast_log statement? I know I am missing something simple ( I hope) I have also followed the instructions on http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html to no avail. I have also copied my app_conference files from another Asterisk box ( it compiles fine on that box). On my new box it will not compile. Obviously, asterisk's API must have changed between the two versions you're using here.. I haven't compiled app_conference against a more recent asterisk than the _old_ stable_1_0 stuff. What version are you compiling against? Also, you can run asterisk under
Re: [Asterisk-Users] app_conference
I don't really like swapping binaries but... I have an app_conference.so binary file I could send to you if you like. It is working on the latest stable cvs as of a few days ago. If you would like it, please let me know and I will get it available. Darren Wiebe [EMAIL PROTECTED] Steve Kann wrote: Shawn Dillon wrote: Thanks to all who have helped me build and test out Asterisk installation thus far. I needed to move my * installation to a new box , due to the fact my test machine would not support PCI 2.2 ( which I am told is required to use my TDM11B). I have * up and running and I am attempting to compile the app_conference source. The MeetMe app has too much echo. I am running Debian 2.4.26 and get tons of compile errors. If I compile right from the CVS of app_conference I get: chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: error: invalid option `abi=altivec' cc1: error: invalid option `dynamic-no-pic' cc1: error: unrecognized option `-faltivec' cc1: error: bad value (7450) for -mcpu= switch cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) make: *** [app_conference.o] Error 1 I then fix the mcpu ( I am on a Pentium4 Box). I comment out the line. I run make clean and make and then get the following. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [conference.o] Error 1 I change my two lines in conference.c as per http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html I run make clean, make and get the following error chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o member.o member.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) member.c: In function `member_exec': member.c:76: error: structure has no member named `dnid' member.c:76: error: structure has no member named `callerid' member.c:76: error: structure has no member named `ani' make: *** [member.o] Error 1 I have edited my member.c to remove any reference to the dnid,callerid and ani , and it compiles. But when someone connects to the conference * crashes. Did you remove the whole ast_log statement? I know I am missing something simple ( I hope) I have also followed the instructions on http://lists.digium.com/pipermail/asterisk-users/2004-September/063765.html to no avail. I have also copied my app_conference files from another Asterisk box ( it compiles fine on that box). On my new
RE: [Asterisk-Users] app_conference
I found this patch a few days ago (on a mailing list), and patched it against the latest cvs which I downloaded for app conference. With these changes I believe everything compiled fine no other tweaks required other then the include dir for asterisk in the make file. On a side note, id like to see a few more features in this module beep on entry part is a big one, however my c/c++ isnt savy enough. I made some attempts but without any luck. If anyone feels bored, try it out and let me know. Donny -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: October 20, 2004 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] app_conference I don't really like swapping binaries but... I have an app_conference.so binary file I could send to you if you like. It is working on the latest stable cvs as of a few days ago. If you would like it, please let me know and I will get it available. Darren Wiebe [EMAIL PROTECTED] Steve Kann wrote: Shawn Dillon wrote: Thanks to all who have helped me build and test out Asterisk installation thus far. I needed to move my * installation to a new box , due to the fact my test machine would not support PCI 2.2 ( which I am told is required to use my TDM11B). I have * up and running and I am attempting to compile the app_conference source. The MeetMe app has too much echo. I am running Debian 2.4.26 and get tons of compile errors. If I compile right from the CVS of app_conference I get: chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: error: invalid option `abi=altivec' cc1: error: invalid option `dynamic-no-pic' cc1: error: unrecognized option `-faltivec' cc1: error: bad value (7450) for -mcpu= switch cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) make: *** [app_conference.o] Error 1 I then fix the mcpu ( I am on a Pentium4 Box). I comment out the line. I run make clean and make and then get the following. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [conference.o] Error 1 I change my two lines in conference.c as per http://lists.digium.com/pipermail/asterisk-users/2004-September/06376 5.html I run make clean, make and get the following error chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o member.o member.c cc1: warning
[Asterisk-Users] app_conference Compile
Hi, I´m just compiling the app_conference but I can´t find the common.h file , those it´s requered. Someone help me to find Common.h file Thanks
