From can_man at gmx.de Fri May 1 04:19:45 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Fri, 01 May 2009 13:19:45 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> Message-ID: <20090501111945.168380@gmx.net> Ciao Giovanni, grazie per la tua risposta. Removing 'hdmi' did make some changes, but it still doesn't work. I have filed a jira: http://jira.freeswitch.org/browse/MODSKYPIAX-33 Buon primo maggio anche a te, Phil -------- Original-Nachricht -------- > Datum: Fri, 1 May 2009 08:20:10 +0200 > Von: Giovanni Maruzzelli > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error > Have a happy MayDay! > > I cannot see the whole mail now, it's clipped for my mobile, but it > seems the nth bizarry of new alsa config file, that creates an hdmi > device even if you do not have one. Try to edit > /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and > delete any mention of 'hdmi'. > If this do not works, please file a jira or write again. > Giovanni > > > > On 5/1/09, Anthony Minessale wrote: > > if you put that info in a jira ticket > > > > http://jira.freeswitch.org > > > > and route it to skypeiax , the guy who maintains that module will see > it. > > > > > > On Thu, Apr 30, 2009 at 5:37 PM, wrote: > > > >> > >> Hello, > >> > >> I am trying to get skypiax working, but I am having trouble with the > >> sound. > >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and > >> I am getting the following error: > >> > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > >> cards.pcm.hdmi > >> > >> > >> I am running centos 5.3 and have followed the installation guide on the > >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When > >> saving > >> the configuration on my desktop I have set the sound card to snd_dummy. > On > >> the server the startup script load snd-dumy like this /sbin/modprobe > >> snd-dummy enable=1. > >> Below is the output of lsmod and the debug output from FS. It would be > >> great if someone could help me fix my problem. > >> > >> Thank you very much. > >> Best wishes, > >> Phil > >> > >> > >> > >> > >> -bash-3.2# lsmod > >> Module Size Used by > >> snd_dummy 12416 0 > >> snd_seq_oss 32832 0 > >> snd_seq_midi_event 7744 1 snd_seq_oss > >> snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event > >> snd_seq_device 7120 1 snd_seq_oss > >> snd_pcm_oss 44480 0 > >> snd_mixer_oss 16512 1 snd_pcm_oss > >> snd_pcm 79624 2 snd_dummy,snd_pcm_oss > >> snd_timer 22088 2 snd_seq,snd_pcm > >> snd 55976 8 > >> > snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > >> soundcore 7456 1 snd > >> snd_page_alloc 8720 1 snd_pcm > >> > >> > >> > >> freeswitch at voipserverServerFreeswitch> load mod_skypiax > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 718 ][none ][-1,-1,-1] > >> globals.debug=0 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 720 ][none ][-1,-1,-1] > >> globals.debug=8 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 731 ][none ][-1,-1,-1] > >> codec-master > >> globals.debug=8 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 734 ][none ][-1,-1,-1] > >> globals.dialplan=XML > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 740 ][none ][-1,-1,-1] > >> globals.context=default > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 743 ][none ][-1,-1,-1] > >> globals.codec_string=gsm,ulaw > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 750 ][none ][-1,-1,-1] > >> globals.codec_rates_string=8000,16000 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 723 ][none ][-1,-1,-1] > >> globals.hold_music= > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 737 ][none ][-1,-1,-1] > >> globals.destination=5000 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] > >> interface_id=1 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] > >> name=skypiax1 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] > Initialized > >> XInitThreads! > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax1 ][-1, 0, 0] > CONFIGURING > >> interface_id=1 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].destination=3101 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax1 ][-1, 0, 0] > >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default > >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > >> 13177[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING > >> interface_id=1 > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > >> 1407 > >> ][skypiax1 ][-1, 0, 0] X Display ':101' opened > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype > >> instance found with id #2097454 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 > >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > 661 > >> ][skypiax1 ][-1, 0, 0] In skypiax_signaling_thread_func: started, > >> p=0x2aaab93226f8 > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||OK||| > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||PROTOCOL 7||| > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||CONNSTATUS ONLINE||| > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||CURRENTUSERHANDLE xyzUK||| > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax1 ][-1, 0, 0] > Skype > >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, > >> cuh: > >> xyzUK, skype_user: xyzUK! > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||USERSTATUS ONLINE||| > >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev > >> 13177[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING > roughly > >> 10 > >> seconds to find a running Skype client and connect to its SKYPE API for > >> interface_id=1 > >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev > >> 13177[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a > running > >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 > >> seconds for CURRENTUSERHANDLE==xyzUK > >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev > >> 13177[(nil)|37 ][WARNINGA 1004 ][skypiax1 ][-1, 0, 0] > Interface_id=1 > >> is now STARTED, the Skype client to which we are connected gave us the > >> correct CURRENTUSERHANDLE (xyzUK) > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 847 ][none ][-1,-1,-1] > >> interface_id=2 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 870 ][none ][-1,-1,-1] > >> name=skypiax2 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 876 ][none ][-1,-1,-1] > Initialized > >> XInitThreads! > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 897 ][skypiax2 ][-1, 0, 0] > CONFIGURING > >> interface_id=2 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 920 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 924 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 > >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 928 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 932 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 935 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 938 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 942 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 946 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].destination=5000 > >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 949 ][skypiax2 ][-1, 0, 0] > >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default > >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > >> 13177[(nil)|37 ][WARNINGA 950 ][skypiax2 ][-1, 0, 0] STARTING > >> interface_id=2 > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > >> 1407 > >> ][skypiax2 ][-1, 0, 0] X Display ':102' opened > >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1309 ][none ][-1,-1,-1] Skype > >> instance found with id #2097454 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 > >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > 661 > >> ][skypiax2 ][-1, 0, 0] In skypiax_signaling_thread_func: started, > >> p=0x2aaab9325c18 > >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] > >> READING: > >> |||OK||| > >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] > >> READING: > >> |||PROTOCOL 7||| > >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] > >> READING: > >> |||CONNSTATUS ONLINE||| > >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] > >> READING: > >> |||CURRENTUSERHANDLE voipserver||| > >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 111 ][skypiax2 ][-1, 0, 0] > Skype > >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, > >> cuh: > >> voipserver, skype_user: voipserver! > >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax2 ][-1, 0, 0] > >> READING: > >> |||USERSTATUS ONLINE||| > >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev > >> 13177[(nil)|37 ][NOTICA 976 ][skypiax2 ][-1, 0, 0] WAITING > roughly > >> 10 > >> seconds to find a running Skype client and connect to its SKYPE API for > >> interface_id=2 > >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev > >> 13177[(nil)|37 ][NOTICA 986 ][skypiax2 ][-1, 0, 0] Found a > running > >> Skype client, connected to its SKYPE API for interface_id=2, waiting 60 > >> seconds for CURRENTUSERHANDLE==voipserver > >> API CALL [load(mod_skypiax)] output: > >> +OK > >> > >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev > >> 13177[(nil)|37 ][WARNINGA 1004 ][skypiax2 ][-1, 0, 0] > Interface_id=2 > >> is now STARTED, the Skype client to which we are connected gave us the > >> correct CURRENTUSERHANDLE (voipserver) > >> > >> > >> > >> > >> > >> > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].interface_id=1 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].context=default > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].dialplan=XML > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].destination=3101 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax1 ][-1, 0, 0] i=1 > >> globals.SKYPIAX_INTERFACES[1].context=default > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1028 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].interface_id=2 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1030 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1032 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1034 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].context=default > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1036 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].dialplan=XML > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1038 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].destination=5000 > >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 1040 ][skypiax2 ][-1, 0, 0] i=2 > >> globals.SKYPIAX_INTERFACES[2].context=default > >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 > >> switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] > >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 > >> switch_loadable_module_process() Adding Endpoint 'skypiax' > >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > >> switch_loadable_module_process() Adding API Function 'sk' > >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > >> switch_loadable_module_process() Adding API Function 'skypiax' > >> freeswitch at voipserverServerFreeswitch> > >> freeswitch at voipserverServerFreeswitch> > >> freeswitch at voipserverServerFreeswitch> > >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] > >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 > >> ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] READING: |||USER paolofun6 > >> PHONE_MOBILE +420775216536||| > >> > >> freeswitch at voipserverServerFreeswitch> > >> freeswitch at voipserverServerFreeswitch> > >> freeswitch at voipserverServerFreeswitch> > >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] > >> switch_channel.c:602 switch_channel_set_name() New Channel > sofia/external/ > >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] > >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() > >> Channel > >> sofia/external/07771236762 at sipgate.co.uk entering state [received][100] > >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() > Remote > >> SDP: > >> v=0 > >> o=root 15141 15141 IN IP4 217.10.66.71 > >> s=session > >> c=IN IP4 217.10.66.71 > >> t=0 0 > >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 > >> a=rtpmap:8 PCMA/8000 > >> a=rtpmap:0 PCMU/8000 > >> a=rtpmap:3 GSM/8000 > >> a=rtpmap:97 iLBC/8000 > >> a=fmtp:97 mode=30 > >> a=rtpmap:18 G729/8000 > >> a=fmtp:18 annexb=no > >> a=rtpmap:112 G726-32/8000 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16 > >> a=silenceSupp:off - - - - > >> a=ptime:20 > >> > >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > sofia_glue_negotiate_sdp() > >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] > >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > sofia_glue_negotiate_sdp() > >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] > >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > sofia_glue_negotiate_sdp() > >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] > >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > sofia_glue_negotiate_sdp() > >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 > sofia_glue_tech_set_codec() > >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 > >> samples > >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 > sofia_glue_negotiate_sdp() > >> Set 2833 dtmf payload to 101 > >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() > >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> > CS_INIT > >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal sofia/external/ > >> 07771236762 at sipgate.co.uk [BREAK] > >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > >> Running State Change CS_INIT > >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> INIT > >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/external/ > >> 07771236762 at sipgate.co.uk SOFIA INIT > >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() > >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> > >> CS_ROUTING > >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal sofia/external/ > >> 07771236762 at sipgate.co.uk [BREAK] > >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> INIT going to sleep > >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > >> Running State Change CS_ROUTING > >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> ROUTING > >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() > >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING > >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 > >> switch_core_standard_on_routing() > >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING > >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > >> Processing 07771236762->00442083324655 in context public > >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing > >> [public->skype_uri] continue=false > >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) > >> [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ > >> break=on-false > >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action > >> bridge(skypiax/skypiax1/xyzTestUK) > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 > >> switch_core_standard_on_routing() (sofia/external/ > >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal sofia/external/ > >> 07771236762 at sipgate.co.uk [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> ROUTING going to sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > >> Running State Change CS_EXECUTE > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> EXECUTE > >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() > >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 > >> switch_core_standard_on_execute() > >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE > >> EXECUTE > >> > sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 > channel_outgoing_channel() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 585 ][ ][-1, 0, 0] > >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 151 ][skypiax1 ][-1, 0, 0] > >> skypiax_codec > >> SUCCESS > >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 > >> switch_channel_set_name() > >> New Channel skypiax/skypiax1/xyzTestUK > >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 773 ][skypiax1 ][-1, 0, 0] Calling > >> Skype, rdest is: xyzTestUK > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > >> ][skypiax1 ][-1, 0, 0] SENDING: |||SET AGC OFF|||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > >> ][skypiax1 ][-1, 0, 0] SENDING: |||SET AEC OFF|||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > >> ][skypiax1 ][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 > channel_outgoing_channel() > >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal > >> skypiax/skypiax1/xyzTestUK [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State > >> Change > >> CS_INIT > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() > >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal > >> skypiax/skypiax1/xyzTestUK [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 182 ][skypiax1 ][-1, 0, 0] > >> skypiax/skypiax1/xyzTestUK CHANNEL INIT > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going > to > >> sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State > >> Change > >> CS_ROUTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 257 ][skypiax1 ][-1, 0, 0] > >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING > >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 > >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change > >> CS_ROUTING > >> -> CS_CONSUME_MEDIA > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal > >> skypiax/skypiax1/xyzTestUK [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 0, 0] > >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING > going > >> to sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State > >> Change > >> CS_CONSUME_MEDIA > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > CONSUME_MEDIA > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||AGC OFF||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||AEC OFF||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 0, 0] > >> READING: > >> |||CALL 455 STATUS UNPLACED||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 0, 0] > Skype > >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: > >> UNPLACED,where: > >> NULL! > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 371 ][skypiax1 ][-1, 3,116] > >> skype_call: 455 is now UNPLACED > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,116] > >> READING: > >> |||CALL 455 STATUS ROUTING||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,116] > Skype > >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: > >> ROUTING,where: > >> NULL! > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 365 ][skypiax1 ][-1, 3,117] > >> skype_call: 455 is now ROUTING > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] > >> READING: > >> |||CALL 455 FAILUREREASON 7||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] > Skype > >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: > >> 7,where: > >> NULL! > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 201 ][skypiax1 ][-1, 3,117] > Skype > >> FAILED on skype_call 455. Let's wait for the FAILED message. > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] > >> READING: > >> |||CALL 455 VAA_INPUT_STATUS FALSE||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] > Skype > >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: > >> FALSE,where: NULL! > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 3,117] > >> READING: > >> |||CALL 455 STATUS FAILED||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 167 ][skypiax1 ][-1, 3,117] > Skype > >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: > FAILED,where: > >> NULL! > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 > >> skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 334 ][skypiax1 ][-1, 3,112] we > >> tried > >> to call Skype on skype_call 455 and Skype has now FAILED > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > 672 > >> ][skypiax1 ][-1, 1,112] skype call ended > >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 > >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK > >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] > >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > >> switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK > >> [KILL] > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 293 ][skypiax1 ][-1, 1,112] > >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal > >> skypiax/skypiax1/xyzTestUK [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] > >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 > >> switch_ivr_originate() Originate Resulted in Error Cause: 16 > >> [NORMAL_CLEARING] > >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() > >> Originate Failed. Cause: NORMAL_CLEARING > >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() > >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > >> switch_channel_perform_hangup() Send signal sofia/external/ > >> 07771236762 at sipgate.co.uk [KILL] > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal sofia/external/ > >> 07771236762 at sipgate.co.uk [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> EXECUTE going to sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > >> Running State Change CS_HANGUP > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> HANGUP > >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: > >> NORMAL_CLEARING > >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() > Responding > >> to > >> INVITE with: 480 > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() > >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: > >> NORMAL_CLEARING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> HANGUP going to sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> Change CS_HANGUP -> CS_REPORTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal sofia/external/ > >> 07771236762 at sipgate.co.uk [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > >> Running State Change CS_REPORTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > >> switch_core_session_reporting_state() (sofia/external/ > >> 07771236762 at sipgate.co.uk) State REPORTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > CONSUME_MEDIA > >> going to sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State > >> Change > >> CS_HANGUP > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 228 ][skypiax1 ][-1, 1,112] hanging > up > >> skype call: 455 > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE 1262 > >> ][skypiax1 ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 235 ][skypiax1 ][-1, 1,112] > >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard > >> HANGUP, > >> cause: NORMAL_CLEARING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP > going > >> to > >> sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change > >> CS_HANGUP -> CS_REPORTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >> switch_core_session_signal_state_change() Send signal > >> skypiax/skypiax1/xyzTestUK [BREAK] > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() > rev > >> 13177[(nil)|37 ][DEBUG_SKYPE 300 ][skypiax1 ][-1, 1,112] > >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State > >> Change > >> CS_REPORTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) > State > >> REPORTING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 > >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) > State > >> REPORTING going to sleep > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change > >> CS_REPORTING -> CS_DESTROY > >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 > >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) > >> Locked, > >> Waiting on external entities > >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 > >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) > Ended > >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 > >> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK > >> [CS_DESTROY] > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State > >> DESTROY > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 > >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard > >> DESTROY > >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State > >> DESTROY going to sleep > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 57 ][skypiax1 ][-1, 1,112] > >> READING: > >> |||ERROR 559 CALL: Action failed||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 91 ][skypiax1 ][-1, 1,112] > Skype > >> got ERROR: |||ERROR||| > >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 > skypiax_signaling_read() > >> rev 13177[(nil)|37 ][DEBUG_SKYPE 93 ][skypiax1 ][-1, 1,110] > >> skype_call now is DOWN > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > 672 > >> ][skypiax1 ][-1, 1,110] skype call ended > >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 > >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE > 687 > >> ][skypiax1 ][-1, 1,110] no session > >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 > >> switch_core_standard_on_reporting() sofia/external/ > >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING > >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 > >> switch_core_session_reporting_state() (sofia/external/ > >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep > >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 > >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) > State > >> Change CS_REPORTING -> CS_DESTROY > >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 > >> switch_core_session_thread() Session 1 (sofia/external/ > >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities > >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 > >> switch_core_session_thread() Session 1 (sofia/external/ > >> 07771236762 at sipgate.co.uk) Ended > >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 > >> switch_core_session_thread() Close Channel sofia/external/ > >> 07771236762 at sipgate.co.uk [CS_DESTROY] > >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > >> switch_core_session_destroy_state() (sofia/external/ > >> 07771236762 at sipgate.co.uk) State DESTROY > >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY > >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 > >> switch_core_standard_on_destroy() > >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY > >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > >> switch_core_session_destroy_state() (sofia/external/ > >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep > >> -- > >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* > >> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > -- > Sent from my mobile device > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a From anthony.minessale at gmail.com Fri May 1 05:31:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 1 May 2009 07:31:31 -0500 Subject: [Freeswitch-users] uuid_displace & FIFO help In-Reply-To: <49FA8D98.3040900@ttnc.co.uk> References: <49F07EC5.5040504@barakatdesigns.net> <191c3a030904231418h6d11e11bp4e84f44ea2abf179@mail.gmail.com> <49F0E3A8.5030400@ttnc.co.uk> <49F16826.5050203@ttnc.co.uk> <6D57020E-7D08-4886-A2BC-6F139E6C1BD6@freeswitch.org> <49F1C880.5040300@ttnc.co.uk> <191c3a030904241859w4bc26e84u3d8640dda76a961e@mail.gmail.com> <49F56F3B.8000906@ttnc.co.uk> <191c3a030904270519h6f85d391p72ca1500f94cfaa5@mail.gmail.com> <49FA8D98.3040900@ttnc.co.uk> Message-ID: <191c3a030905010531j7731edc1xdb312a5aef2153ec@mail.gmail.com> can you submit the patch over jira http://jira.freeswitch.org they do not transfer well over email and we need to document all the patches. On Fri, May 1, 2009 at 12:50 AM, TTNC - Adnan Barakat wrote: > Anthony Minessale wrote: > >> Also is there any way to stop uuid_broadcast as I'd >> need to stop it somehow if the destination picks up? >> >> break all >> > "uuid_broadcast phrase::saynumber,1" doesn't set the > 'current_application_response' variable in the same way as "uuid_broadcast > playback::filename.wav" does (which my script looks for to know when > to move on to the next application). > > I've attached a patch which sets this variable if it's any use to anyone > (I'm not that great at C so I hope it's correct, any comments/improvements > are welcome). > > > Thanks again > > Adnan > > Index: src/mod/applications/mod_dptools/mod_dptools.c > =================================================================== > --- src/mod/applications/mod_dptools/mod_dptools.c (revision 13172) > +++ src/mod/applications/mod_dptools/mod_dptools.c (working copy) > @@ -1807,6 +1807,7 @@ > char *mydata = NULL; > switch_input_args_t args = { 0 }; > switch_channel_t *channel = > switch_core_session_get_channel(session); > + switch_status_t status; > > if (!switch_strlen_zero(data) && (mydata = > switch_core_session_strdup(session, data))) { > const char *lang; > @@ -1825,8 +1826,23 @@ > > switch_channel_set_variable(channel, > SWITCH_PLAYBACK_TERMINATOR_USED, "" ); > > - switch_ivr_phrase_macro(session, macro, mdata, lang, > &args); > + status = switch_ivr_phrase_macro(session, macro, mdata, > lang, &args); > + } else { > + status = SWITCH_STATUS_NOOP; > } > + > + switch (status) { > + case SWITCH_STATUS_SUCCESS: > + case SWITCH_STATUS_BREAK: > + switch_channel_set_variable(channel, > SWITCH_CURRENT_APPLICATION_RESPONSE_VARIABLE, "PHRASE PLAYED"); > + break; > + case SWITCH_STATUS_NOOP: > + switch_channel_set_variable(channel, > SWITCH_CURRENT_APPLICATION_RESPONSE_VARIABLE, "NOTHING"); > + break; > + default: > + switch_channel_set_variable(channel, > SWITCH_CURRENT_APPLICATION_RESPONSE_VARIABLE, "UNKNOWN ERROR"); > + break; > + } > } > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/9809ce3f/attachment.html From moizchinoy at gmail.com Fri May 1 06:38:52 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Fri, 1 May 2009 18:38:52 +0500 Subject: [Freeswitch-users] Ways of Integrating Sphinx... Message-ID: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com> Hi, I know only two ways of Sphinx - FS integration and its through mod_pocketsphinx and ESL. Performance with mod_pocketsphinx was not very good especially prompts were not playing properly. I haven't tried ESL. Can anyone guide what are other possibilities and which one is best in stability and can any one be deployed in live environment. -- Regards, Moiz Chinoy. From brian at freeswitch.org Fri May 1 06:54:34 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 1 May 2009 08:54:34 -0500 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com> Message-ID: Can you elaborate more on this? I use it often in testing with the Pizza Demo..... It works fine. What SVN rev are you on also? /b On May 1, 2009, at 8:38 AM, Moiz Chinoy wrote: > Performance with mod_pocketsphinx was not very good especially prompts > were not playing properly. I haven't tried ESL. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/92a221f4/attachment.html From gmaruzz at celliax.org Fri May 1 07:14:12 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 1 May 2009 16:14:12 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <20090501111945.168380@gmx.net> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> <20090501111945.168380@gmx.net> Message-ID: <7b197bef0905010714l4fc38792o63877627704c1939@mail.gmail.com> Gruss Phil, actually it was shooting in the dark from my side, because I not yet tested centos5.3, only centos5.2 As soon as I test it out I'll be back to you. Thanks for filing the Jira. -giovanni On Fri, May 1, 2009 at 1:19 PM, wrote: > Ciao Giovanni, > > grazie per la tua risposta. Removing 'hdmi' did make some changes, but it > still doesn't work. I have filed a jira: > > http://jira.freeswitch.org/browse/MODSKYPIAX-33 > > Buon primo maggio anche a te, > Phil > > -------- Original-Nachricht -------- >> Datum: Fri, 1 May 2009 08:20:10 +0200 >> Von: Giovanni Maruzzelli >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O ? ?error > >> Have a happy MayDay! >> >> I cannot see the whole mail now, it's clipped for my mobile, but it >> seems the nth bizarry of new alsa config file, that creates an hdmi >> device even if you do not have one. Try to edit >> /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and >> delete any mention of 'hdmi'. >> If this do not works, please file a jira or write again. >> Giovanni >> >> >> >> On 5/1/09, Anthony Minessale wrote: >> > if you put that info in a jira ticket >> > >> > http://jira.freeswitch.org >> > >> > and route it to skypeiax , the guy who maintains that module will see >> it. >> > >> > >> > On Thu, Apr 30, 2009 at 5:37 PM, wrote: >> > >> >> >> >> Hello, >> >> >> >> I am trying to get skypiax working, but I am having trouble with the >> >> sound. >> >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and >> >> I am getting the following error: >> >> >> >> ? ? ? ?ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM >> >> cards.pcm.hdmi >> >> >> >> >> >> I am running centos 5.3 and have followed the installation guide on the >> >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When >> >> saving >> >> the configuration on my desktop I have set the sound card to snd_dummy. >> On >> >> the server the startup script load snd-dumy like this /sbin/modprobe >> >> snd-dummy enable=1. >> >> Below is the output of lsmod and the debug output from FS. It would be >> >> great if someone could help me fix my problem. >> >> >> >> Thank you very much. >> >> Best wishes, >> >> Phil >> >> >> >> >> >> >> >> >> >> -bash-3.2# lsmod >> >> Module ? ? ? ? ? ? ? ? ?Size ?Used by >> >> snd_dummy ? ? ? ? ? ? ?12416 ?0 >> >> snd_seq_oss ? ? ? ? ? ?32832 ?0 >> >> snd_seq_midi_event ? ? ?7744 ?1 snd_seq_oss >> >> snd_seq ? ? ? ? ? ? ? ?55200 ?4 snd_seq_oss,snd_seq_midi_event >> >> snd_seq_device ? ? ? ? ?7120 ?1 snd_seq_oss >> >> snd_pcm_oss ? ? ? ? ? ?44480 ?0 >> >> snd_mixer_oss ? ? ? ? ?16512 ?1 snd_pcm_oss >> >> snd_pcm ? ? ? ? ? ? ? ?79624 ?2 snd_dummy,snd_pcm_oss >> >> snd_timer ? ? ? ? ? ? ?22088 ?2 snd_seq,snd_pcm >> >> snd ? ? ? ? ? ? ? ? ? ?55976 ?8 >> >> >> snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer >> >> soundcore ? ? ? ? ? ? ? 7456 ?1 snd >> >> snd_page_alloc ? ? ? ? ?8720 ?1 snd_pcm >> >> >> >> >> >> >> >> freeswitch at voipserverServerFreeswitch> load mod_skypiax >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?718 ?][none ? ? ?][-1,-1,-1] >> >> globals.debug=0 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?720 ?][none ? ? ?][-1,-1,-1] >> >> globals.debug=8 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?731 ?][none ? ? ?][-1,-1,-1] >> >> codec-master >> >> globals.debug=8 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?734 ?][none ? ? ?][-1,-1,-1] >> >> globals.dialplan=XML >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?740 ?][none ? ? ?][-1,-1,-1] >> >> globals.context=default >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?743 ?][none ? ? ?][-1,-1,-1] >> >> globals.codec_string=gsm,ulaw >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?750 ?][none ? ? ?][-1,-1,-1] >> >> globals.codec_rates_string=8000,16000 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?723 ?][none ? ? ?][-1,-1,-1] >> >> globals.hold_music= >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?737 ?][none ? ? ?][-1,-1,-1] >> >> globals.destination=5000 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? ?][-1,-1,-1] >> >> interface_id=1 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? ?][-1,-1,-1] >> >> name=skypiax1 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? ?][-1,-1,-1] >> Initialized >> >> XInitThreads! >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax1 ?][-1, 0, 0] >> CONFIGURING >> >> interface_id=1 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 >> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 >> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 >> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 >> globals.SKYPIAX_INTERFACES[interface_id].destination=3101 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax1 ?][-1, 0, 0] >> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax1 ?][-1, 0, 0] STARTING >> >> interface_id=1 >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> >> 1407 >> >> ][skypiax1 ?][-1, 0, 0] X Display ':101' opened >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? ?][-1,-1,-1] Skype >> >> instance found with id #2097454 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 >> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> 661 >> >> ?][skypiax1 ?][-1, 0, 0] In skypiax_signaling_thread_func: started, >> >> p=0x2aaab93226f8 >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||OK||| >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||PROTOCOL 7||| >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||CONNSTATUS ONLINE||| >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||CURRENTUSERHANDLE xyzUK||| >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax1 ?][-1, 0, 0] >> Skype >> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >> >> cuh: >> >> xyzUK, skype_user: xyzUK! >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||USERSTATUS ONLINE||| >> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev >> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax1 ?][-1, 0, 0] WAITING >> roughly >> >> 10 >> >> seconds to find a running Skype client and connect to its SKYPE API for >> >> interface_id=1 >> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev >> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax1 ?][-1, 0, 0] Found a >> running >> >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >> >> seconds for CURRENTUSERHANDLE==xyzUK >> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev >> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax1 ?][-1, 0, 0] >> Interface_id=1 >> >> is now STARTED, the Skype client to which we are connected gave us the >> >> correct CURRENTUSERHANDLE (xyzUK) >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? ?][-1,-1,-1] >> >> interface_id=2 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? ?][-1,-1,-1] >> >> name=skypiax2 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? ?][-1,-1,-1] >> Initialized >> >> XInitThreads! >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax2 ?][-1, 0, 0] >> CONFIGURING >> >> interface_id=2 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 >> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 >> >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 >> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 >> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 >> globals.SKYPIAX_INTERFACES[interface_id].destination=5000 >> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax2 ?][-1, 0, 0] >> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax2 ?][-1, 0, 0] STARTING >> >> interface_id=2 >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> >> 1407 >> >> ][skypiax2 ?][-1, 0, 0] X Display ':102' opened >> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? ?][-1,-1,-1] Skype >> >> instance found with id #2097454 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 >> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> 661 >> >> ?][skypiax2 ?][-1, 0, 0] In skypiax_signaling_thread_func: started, >> >> p=0x2aaab9325c18 >> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >> >> READING: >> >> |||OK||| >> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >> >> READING: >> >> |||PROTOCOL 7||| >> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >> >> READING: >> >> |||CONNSTATUS ONLINE||| >> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >> >> READING: >> >> |||CURRENTUSERHANDLE voipserver||| >> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax2 ?][-1, 0, 0] >> Skype >> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >> >> cuh: >> >> voipserver, skype_user: voipserver! >> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >> >> READING: >> >> |||USERSTATUS ONLINE||| >> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev >> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax2 ?][-1, 0, 0] WAITING >> roughly >> >> 10 >> >> seconds to find a running Skype client and connect to its SKYPE API for >> >> interface_id=2 >> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev >> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax2 ?][-1, 0, 0] Found a >> running >> >> Skype client, connected to its SKYPE API for interface_id=2, waiting 60 >> >> seconds for CURRENTUSERHANDLE==voipserver >> >> API CALL [load(mod_skypiax)] output: >> >> +OK >> >> >> >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev >> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax2 ?][-1, 0, 0] >> Interface_id=2 >> >> is now STARTED, the Skype client to which we are connected gave us the >> >> correct CURRENTUSERHANDLE (voipserver) >> >> >> >> >> >> >> >> >> >> >> >> >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].interface_id=1 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].context=default >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].dialplan=XML >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].destination=3101 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax1 ?][-1, 0, 0] i=1 >> >> globals.SKYPIAX_INTERFACES[1].context=default >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].interface_id=2 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].context=default >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].dialplan=XML >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].destination=5000 >> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax2 ?][-1, 0, 0] i=2 >> >> globals.SKYPIAX_INTERFACES[2].context=default >> >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 >> >> switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] >> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 >> >> switch_loadable_module_process() Adding Endpoint 'skypiax' >> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >> >> switch_loadable_module_process() Adding API Function 'sk' >> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >> >> switch_loadable_module_process() Adding API Function 'skypiax' >> >> freeswitch at voipserverServerFreeswitch> >> >> freeswitch at voipserverServerFreeswitch> >> >> freeswitch at voipserverServerFreeswitch> >> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] >> >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 >> >> ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] READING: |||USER paolofun6 >> >> PHONE_MOBILE +420775216536||| >> >> >> >> freeswitch at voipserverServerFreeswitch> >> >> freeswitch at voipserverServerFreeswitch> >> >> freeswitch at voipserverServerFreeswitch> >> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] >> >> switch_channel.c:602 switch_channel_set_name() New Channel >> sofia/external/ >> >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] >> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() >> >> Channel >> >> sofia/external/07771236762 at sipgate.co.uk entering state [received][100] >> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() >> Remote >> >> SDP: >> >> v=0 >> >> o=root 15141 15141 IN IP4 217.10.66.71 >> >> s=session >> >> c=IN IP4 217.10.66.71 >> >> t=0 0 >> >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:0 PCMU/8000 >> >> a=rtpmap:3 GSM/8000 >> >> a=rtpmap:97 iLBC/8000 >> >> a=fmtp:97 mode=30 >> >> a=rtpmap:18 G729/8000 >> >> a=fmtp:18 annexb=no >> >> a=rtpmap:112 G726-32/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:20 >> >> >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >> sofia_glue_negotiate_sdp() >> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >> sofia_glue_negotiate_sdp() >> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >> sofia_glue_negotiate_sdp() >> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >> sofia_glue_negotiate_sdp() >> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 >> sofia_glue_tech_set_codec() >> >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 >> >> samples >> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 >> sofia_glue_negotiate_sdp() >> >> Set 2833 dtmf payload to 101 >> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() >> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> >> CS_INIT >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal sofia/external/ >> >> 07771236762 at sipgate.co.uk [BREAK] >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> >> Running State Change CS_INIT >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> INIT >> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() >> sofia/external/ >> >> 07771236762 at sipgate.co.uk SOFIA INIT >> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() >> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> >> >> CS_ROUTING >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal sofia/external/ >> >> 07771236762 at sipgate.co.uk [BREAK] >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> INIT going to sleep >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> >> Running State Change CS_ROUTING >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> ROUTING >> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING >> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 >> >> switch_core_standard_on_routing() >> >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING >> >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> >> Processing 07771236762->00442083324655 in context public >> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing >> >> [public->skype_uri] continue=false >> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) >> >> [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ >> >> break=on-false >> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action >> >> bridge(skypiax/skypiax1/xyzTestUK) >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 >> >> switch_core_standard_on_routing() (sofia/external/ >> >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal sofia/external/ >> >> 07771236762 at sipgate.co.uk [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> ROUTING going to sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> >> Running State Change CS_EXECUTE >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> EXECUTE >> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() >> >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 >> >> switch_core_standard_on_execute() >> >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE >> >> EXECUTE >> >> >> sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 >> channel_outgoing_channel() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?585 ?][ ? ? ? ? ?][-1, 0, 0] >> >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?151 ?][skypiax1 ?][-1, 0, 0] >> >> skypiax_codec >> >> SUCCESS >> >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 >> >> switch_channel_set_name() >> >> New Channel skypiax/skypiax1/xyzTestUK >> >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?773 ?][skypiax1 ?][-1, 0, 0] Calling >> >> Skype, rdest is: xyzTestUK >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AGC OFF|||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AEC OFF|||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 >> channel_outgoing_channel() >> >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal >> >> skypiax/skypiax1/xyzTestUK [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> >> Change >> >> CS_INIT >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() >> >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal >> >> skypiax/skypiax1/xyzTestUK [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?182 ?][skypiax1 ?][-1, 0, 0] >> >> skypiax/skypiax1/xyzTestUK CHANNEL INIT >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going >> to >> >> sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> >> Change >> >> CS_ROUTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?257 ?][skypiax1 ?][-1, 0, 0] >> >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 >> >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change >> >> CS_ROUTING >> >> -> CS_CONSUME_MEDIA >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal >> >> skypiax/skypiax1/xyzTestUK [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >> going >> >> to sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> >> Change >> >> CS_CONSUME_MEDIA >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State >> CONSUME_MEDIA >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||AGC OFF||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||AEC OFF||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >> >> READING: >> >> |||CALL 455 STATUS UNPLACED||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 0, 0] >> Skype >> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> >> UNPLACED,where: >> >> NULL! >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?371 ?][skypiax1 ?][-1, 3,116] >> >> skype_call: 455 is now UNPLACED >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,116] >> >> READING: >> >> |||CALL 455 STATUS ROUTING||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,116] >> Skype >> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> >> ROUTING,where: >> >> NULL! >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?365 ?][skypiax1 ?][-1, 3,117] >> >> skype_call: 455 is now ROUTING >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >> >> READING: >> >> |||CALL 455 FAILUREREASON 7||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >> Skype >> >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: >> >> 7,where: >> >> NULL! >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?201 ?][skypiax1 ?][-1, 3,117] >> Skype >> >> FAILED on skype_call 455. Let's wait for the FAILED message. >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >> >> READING: >> >> |||CALL 455 VAA_INPUT_STATUS FALSE||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >> Skype >> >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: >> >> FALSE,where: NULL! >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >> >> READING: >> >> |||CALL 455 STATUS FAILED||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >> Skype >> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >> FAILED,where: >> >> NULL! >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 >> >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?334 ?][skypiax1 ?][-1, 3,112] we >> >> tried >> >> to call Skype on skype_call 455 and Skype has now FAILED >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> 672 >> >> ?][skypiax1 ?][-1, 1,112] skype call ended >> >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 >> >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK >> >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] >> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >> >> switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK >> >> [KILL] >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?293 ?][skypiax1 ?][-1, 1,112] >> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal >> >> skypiax/skypiax1/xyzTestUK [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 1,112] >> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 >> >> switch_ivr_originate() Originate Resulted in Error Cause: 16 >> >> [NORMAL_CLEARING] >> >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() >> >> Originate Failed. ?Cause: NORMAL_CLEARING >> >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() >> >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] >> >> [NORMAL_CLEARING] >> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >> >> switch_channel_perform_hangup() Send signal sofia/external/ >> >> 07771236762 at sipgate.co.uk [KILL] >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal sofia/external/ >> >> 07771236762 at sipgate.co.uk [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> EXECUTE going to sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> >> Running State Change CS_HANGUP >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> HANGUP >> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >> >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: >> >> NORMAL_CLEARING >> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >> Responding >> >> to >> >> INVITE with: 480 >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >> >> switch_core_standard_on_hangup() >> >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: >> >> NORMAL_CLEARING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> HANGUP going to sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> Change CS_HANGUP -> CS_REPORTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal sofia/external/ >> >> 07771236762 at sipgate.co.uk [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> >> Running State Change CS_REPORTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> >> switch_core_session_reporting_state() (sofia/external/ >> >> 07771236762 at sipgate.co.uk) State REPORTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State >> CONSUME_MEDIA >> >> going to sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> >> Change >> >> CS_HANGUP >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?228 ?][skypiax1 ?][-1, 1,112] hanging >> up >> >> skype call: 455 >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >> >> ][skypiax1 ?][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?235 ?][skypiax1 ?][-1, 1,112] >> >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >> >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard >> >> HANGUP, >> >> cause: NORMAL_CLEARING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >> going >> >> to >> >> sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >> >> CS_HANGUP -> CS_REPORTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >> >> switch_core_session_signal_state_change() Send signal >> >> skypiax/skypiax1/xyzTestUK [BREAK] >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >> rev >> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 1,112] >> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >> >> Change >> >> CS_REPORTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) >> State >> >> REPORTING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 >> >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard >> >> REPORTING, cause: NORMAL_CLEARING >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) >> State >> >> REPORTING going to sleep >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >> >> CS_REPORTING -> CS_DESTROY >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 >> >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >> >> Locked, >> >> Waiting on external entities >> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 >> >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >> Ended >> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 >> >> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK >> >> [CS_DESTROY] >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >> >> DESTROY >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 >> >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard >> >> DESTROY >> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >> >> DESTROY going to sleep >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 1,112] >> >> READING: >> >> |||ERROR 559 CALL: Action failed||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?91 ? ][skypiax1 ?][-1, 1,112] >> Skype >> >> got ERROR: |||ERROR||| >> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 >> skypiax_signaling_read() >> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?93 ? ][skypiax1 ?][-1, 1,110] >> >> skype_call now is DOWN >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> 672 >> >> ?][skypiax1 ?][-1, 1,110] skype call ended >> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 >> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >> 687 >> >> ?][skypiax1 ?][-1, 1,110] no session >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 >> >> switch_core_standard_on_reporting() sofia/external/ >> >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 >> >> switch_core_session_reporting_state() (sofia/external/ >> >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 >> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >> State >> >> Change CS_REPORTING -> CS_DESTROY >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 >> >> switch_core_session_thread() Session 1 (sofia/external/ >> >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities >> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 >> >> switch_core_session_thread() Session 1 (sofia/external/ >> >> 07771236762 at sipgate.co.uk) Ended >> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 >> >> switch_core_session_thread() Close Channel sofia/external/ >> >> 07771236762 at sipgate.co.uk [CS_DESTROY] >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >> >> switch_core_session_destroy_state() (sofia/external/ >> >> 07771236762 at sipgate.co.uk) State DESTROY >> >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >> >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 >> >> switch_core_standard_on_destroy() >> >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY >> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >> >> switch_core_session_destroy_state() (sofia/external/ >> >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep >> >> -- >> >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + >> >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* >> >> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at 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 at hotmail.com >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> > pstn:213-799-1400 >> > >> >> -- >> Sent from my mobile device >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Fri May 1 07:43:54 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 1 May 2009 10:43:54 -0400 Subject: [Freeswitch-users] Latest SVN update gives Windows Express compiler errors ... In-Reply-To: <8CB98298639AEA6-280-33C3@webmail-dx08.sysops.aol.com> References: <8CB98298639AEA6-280-33C3@webmail-dx08.sysops.aol.com> Message-ID: <5BFA959A-5517-43BC-BA22-205791AD659B@jerris.com> Do you have any specifics of the errors? Mike On May 1, 2009, at 12:07 AM, mszlazak at aol.com wrote: > I'm getting Windows Express compiler errors on the latest svn update > to trunk 13213. > It looks like the path is wrong to some files. > Instead of folder "Debug", it's looking for files in folder "Debug > DLL" > > Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/d200f75e/attachment.html From gustavodartagnan at yahoo.com Fri May 1 10:30:12 2009 From: gustavodartagnan at yahoo.com (Gustavo Dartagnan Xavier) Date: Fri, 1 May 2009 10:30:12 -0700 (PDT) Subject: [Freeswitch-users] Migrating Asterisk Realtime to Freeswitch Message-ID: <779835.31950.qm@web57102.mail.re3.yahoo.com> Hello, Does anyone know if there is an easy way to migrate from a cluster of asterisk realtime (n Asterisk's boxes connected to an Database) to freeswitch? I was thinking to use an web application like Glassfish/Oracle Application to deal with database requests an generate all information in realtime to Freeswitch through XML Curl, do you think this is the best way to do it? Thanks in advance. Regards, Gustavo Dartagnan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/4eb45aeb/attachment.html From msc at freeswitch.org Fri May 1 11:19:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 1 May 2009 11:19:22 -0700 Subject: [Freeswitch-users] Migrating Asterisk Realtime to Freeswitch In-Reply-To: <779835.31950.qm@web57102.mail.re3.yahoo.com> References: <779835.31950.qm@web57102.mail.re3.yahoo.com> Message-ID: <87f2f3b90905011119q28c246e7p4eb9accf2a0cadc5@mail.gmail.com> On Fri, May 1, 2009 at 10:30 AM, Gustavo Dartagnan Xavier < gustavodartagnan at yahoo.com> wrote: > Hello, > > Does anyone know if there is an easy way to migrate from a cluster of > asterisk realtime (n Asterisk's boxes connected to an Database) to > freeswitch? > > I was thinking to use an web application like Glassfish/Oracle Application > to deal with database requests an generate all information in realtime to > Freeswitch through XML Curl, do you think this is the best way to do it? > I'm sure the mod_xml_curl is the right way in FreeSWITCH, but I don't know about your backend db stuff. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/84e327d5/attachment.html From gmaruzz at celliax.org Fri May 1 12:25:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 1 May 2009 21:25:43 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <7b197bef0905010714l4fc38792o63877627704c1939@mail.gmail.com> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> <20090501111945.168380@gmx.net> <7b197bef0905010714l4fc38792o63877627704c1939@mail.gmail.com> Message-ID: <7b197bef0905011225t525dc47cu2c3f8c9b548e4600@mail.gmail.com> Hi Phil, I just tried all the steps (exactly, just cut and paste) from the wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS_5.2.2C_from_scratch I substituted 5.3 instead of 5.2. I'm afraid it worked flawlessly for me. (shocked about: Anthony is right about CentOS being "boring and predictable", good qualities for a server OS!) At the start of Skype clients it will tell bizarre things about hdmi, but they are unharmful (I've not edited the alsa stuff, it still groak about non-existent hdmi, but it works nonetheless). So, I suspect your problems have some other cause. Now I go read the Jira and the attached files, and I hope to be more of help. -giovanni On Fri, May 1, 2009 at 4:14 PM, Giovanni Maruzzelli wrote: > Gruss Phil, > > actually it was shooting in the dark from my side, because I not yet > tested centos5.3, only centos5.2 > > As soon as I test it out I'll be back to you. > Thanks for filing the Jira. > > -giovanni > > > On Fri, May 1, 2009 at 1:19 PM, ? wrote: >> Ciao Giovanni, >> >> grazie per la tua risposta. Removing 'hdmi' did make some changes, but it >> still doesn't work. I have filed a jira: >> >> http://jira.freeswitch.org/browse/MODSKYPIAX-33 >> >> Buon primo maggio anche a te, >> Phil >> >> -------- Original-Nachricht -------- >>> Datum: Fri, 1 May 2009 08:20:10 +0200 >>> Von: Giovanni Maruzzelli >>> An: freeswitch-users at lists.freeswitch.org >>> Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O ? ?error >> >>> Have a happy MayDay! >>> >>> I cannot see the whole mail now, it's clipped for my mobile, but it >>> seems the nth bizarry of new alsa config file, that creates an hdmi >>> device even if you do not have one. Try to edit >>> /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and >>> delete any mention of 'hdmi'. >>> If this do not works, please file a jira or write again. >>> Giovanni >>> >>> >>> >>> On 5/1/09, Anthony Minessale wrote: >>> > if you put that info in a jira ticket >>> > >>> > http://jira.freeswitch.org >>> > >>> > and route it to skypeiax , the guy who maintains that module will see >>> it. >>> > >>> > >>> > On Thu, Apr 30, 2009 at 5:37 PM, wrote: >>> > >>> >> >>> >> Hello, >>> >> >>> >> I am trying to get skypiax working, but I am having trouble with the >>> >> sound. >>> >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and >>> >> I am getting the following error: >>> >> >>> >> ? ? ? ?ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM >>> >> cards.pcm.hdmi >>> >> >>> >> >>> >> I am running centos 5.3 and have followed the installation guide on the >>> >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When >>> >> saving >>> >> the configuration on my desktop I have set the sound card to snd_dummy. >>> On >>> >> the server the startup script load snd-dumy like this /sbin/modprobe >>> >> snd-dummy enable=1. >>> >> Below is the output of lsmod and the debug output from FS. It would be >>> >> great if someone could help me fix my problem. >>> >> >>> >> Thank you very much. >>> >> Best wishes, >>> >> Phil >>> >> >>> >> >>> >> >>> >> >>> >> -bash-3.2# lsmod >>> >> Module ? ? ? ? ? ? ? ? ?Size ?Used by >>> >> snd_dummy ? ? ? ? ? ? ?12416 ?0 >>> >> snd_seq_oss ? ? ? ? ? ?32832 ?0 >>> >> snd_seq_midi_event ? ? ?7744 ?1 snd_seq_oss >>> >> snd_seq ? ? ? ? ? ? ? ?55200 ?4 snd_seq_oss,snd_seq_midi_event >>> >> snd_seq_device ? ? ? ? ?7120 ?1 snd_seq_oss >>> >> snd_pcm_oss ? ? ? ? ? ?44480 ?0 >>> >> snd_mixer_oss ? ? ? ? ?16512 ?1 snd_pcm_oss >>> >> snd_pcm ? ? ? ? ? ? ? ?79624 ?2 snd_dummy,snd_pcm_oss >>> >> snd_timer ? ? ? ? ? ? ?22088 ?2 snd_seq,snd_pcm >>> >> snd ? ? ? ? ? ? ? ? ? ?55976 ?8 >>> >> >>> snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer >>> >> soundcore ? ? ? ? ? ? ? 7456 ?1 snd >>> >> snd_page_alloc ? ? ? ? ?8720 ?1 snd_pcm >>> >> >>> >> >>> >> >>> >> freeswitch at voipserverServerFreeswitch> load mod_skypiax >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?718 ?][none ? ? ?][-1,-1,-1] >>> >> globals.debug=0 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?720 ?][none ? ? ?][-1,-1,-1] >>> >> globals.debug=8 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?731 ?][none ? ? ?][-1,-1,-1] >>> >> codec-master >>> >> globals.debug=8 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?734 ?][none ? ? ?][-1,-1,-1] >>> >> globals.dialplan=XML >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?740 ?][none ? ? ?][-1,-1,-1] >>> >> globals.context=default >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?743 ?][none ? ? ?][-1,-1,-1] >>> >> globals.codec_string=gsm,ulaw >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?750 ?][none ? ? ?][-1,-1,-1] >>> >> globals.codec_rates_string=8000,16000 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?723 ?][none ? ? ?][-1,-1,-1] >>> >> globals.hold_music= >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?737 ?][none ? ? ?][-1,-1,-1] >>> >> globals.destination=5000 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? ?][-1,-1,-1] >>> >> interface_id=1 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? ?][-1,-1,-1] >>> >> name=skypiax1 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? ?][-1,-1,-1] >>> Initialized >>> >> XInitThreads! >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax1 ?][-1, 0, 0] >>> CONFIGURING >>> >> interface_id=1 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 >>> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 >>> globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 >>> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 >>> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 >>> globals.SKYPIAX_INTERFACES[interface_id].destination=3101 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax1 ?][-1, 0, 0] >>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >>> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax1 ?][-1, 0, 0] STARTING >>> >> interface_id=1 >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >>> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> >> 1407 >>> >> ][skypiax1 ?][-1, 0, 0] X Display ':101' opened >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? ?][-1,-1,-1] Skype >>> >> instance found with id #2097454 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 >>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> 661 >>> >> ?][skypiax1 ?][-1, 0, 0] In skypiax_signaling_thread_func: started, >>> >> p=0x2aaab93226f8 >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||OK||| >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||PROTOCOL 7||| >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||CONNSTATUS ONLINE||| >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||CURRENTUSERHANDLE xyzUK||| >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax1 ?][-1, 0, 0] >>> Skype >>> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >>> >> cuh: >>> >> xyzUK, skype_user: xyzUK! >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||USERSTATUS ONLINE||| >>> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev >>> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax1 ?][-1, 0, 0] WAITING >>> roughly >>> >> 10 >>> >> seconds to find a running Skype client and connect to its SKYPE API for >>> >> interface_id=1 >>> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev >>> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax1 ?][-1, 0, 0] Found a >>> running >>> >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >>> >> seconds for CURRENTUSERHANDLE==xyzUK >>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev >>> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax1 ?][-1, 0, 0] >>> Interface_id=1 >>> >> is now STARTED, the Skype client to which we are connected gave us the >>> >> correct CURRENTUSERHANDLE (xyzUK) >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? ?][-1,-1,-1] >>> >> interface_id=2 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? ?][-1,-1,-1] >>> >> name=skypiax2 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? ?][-1,-1,-1] >>> Initialized >>> >> XInitThreads! >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax2 ?][-1, 0, 0] >>> CONFIGURING >>> >> interface_id=2 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 >>> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 >>> >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 >>> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 >>> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 >>> globals.SKYPIAX_INTERFACES[interface_id].destination=5000 >>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax2 ?][-1, 0, 0] >>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >>> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax2 ?][-1, 0, 0] STARTING >>> >> interface_id=2 >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >>> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> >> 1407 >>> >> ][skypiax2 ?][-1, 0, 0] X Display ':102' opened >>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? ?][-1,-1,-1] Skype >>> >> instance found with id #2097454 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 >>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> 661 >>> >> ?][skypiax2 ?][-1, 0, 0] In skypiax_signaling_thread_func: started, >>> >> p=0x2aaab9325c18 >>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>> >> READING: >>> >> |||OK||| >>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>> >> READING: >>> >> |||PROTOCOL 7||| >>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>> >> READING: >>> >> |||CONNSTATUS ONLINE||| >>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>> >> READING: >>> >> |||CURRENTUSERHANDLE voipserver||| >>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax2 ?][-1, 0, 0] >>> Skype >>> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >>> >> cuh: >>> >> voipserver, skype_user: voipserver! >>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>> >> READING: >>> >> |||USERSTATUS ONLINE||| >>> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev >>> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax2 ?][-1, 0, 0] WAITING >>> roughly >>> >> 10 >>> >> seconds to find a running Skype client and connect to its SKYPE API for >>> >> interface_id=2 >>> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev >>> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax2 ?][-1, 0, 0] Found a >>> running >>> >> Skype client, connected to its SKYPE API for interface_id=2, waiting 60 >>> >> seconds for CURRENTUSERHANDLE==voipserver >>> >> API CALL [load(mod_skypiax)] output: >>> >> +OK >>> >> >>> >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev >>> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax2 ?][-1, 0, 0] >>> Interface_id=2 >>> >> is now STARTED, the Skype client to which we are connected gave us the >>> >> correct CURRENTUSERHANDLE (voipserver) >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].interface_id=1 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].context=default >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].dialplan=XML >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].destination=3101 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax1 ?][-1, 0, 0] i=1 >>> >> globals.SKYPIAX_INTERFACES[1].context=default >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].interface_id=2 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].context=default >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].dialplan=XML >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].destination=5000 >>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax2 ?][-1, 0, 0] i=2 >>> >> globals.SKYPIAX_INTERFACES[2].context=default >>> >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 >>> >> switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] >>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 >>> >> switch_loadable_module_process() Adding Endpoint 'skypiax' >>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >>> >> switch_loadable_module_process() Adding API Function 'sk' >>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >>> >> switch_loadable_module_process() Adding API Function 'skypiax' >>> >> freeswitch at voipserverServerFreeswitch> >>> >> freeswitch at voipserverServerFreeswitch> >>> >> freeswitch at voipserverServerFreeswitch> >>> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] >>> >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 >>> >> ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] READING: |||USER paolofun6 >>> >> PHONE_MOBILE +420775216536||| >>> >> >>> >> freeswitch at voipserverServerFreeswitch> >>> >> freeswitch at voipserverServerFreeswitch> >>> >> freeswitch at voipserverServerFreeswitch> >>> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] >>> >> switch_channel.c:602 switch_channel_set_name() New Channel >>> sofia/external/ >>> >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] >>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() >>> >> Channel >>> >> sofia/external/07771236762 at sipgate.co.uk entering state [received][100] >>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() >>> Remote >>> >> SDP: >>> >> v=0 >>> >> o=root 15141 15141 IN IP4 217.10.66.71 >>> >> s=session >>> >> c=IN IP4 217.10.66.71 >>> >> t=0 0 >>> >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 >>> >> a=rtpmap:8 PCMA/8000 >>> >> a=rtpmap:0 PCMU/8000 >>> >> a=rtpmap:3 GSM/8000 >>> >> a=rtpmap:97 iLBC/8000 >>> >> a=fmtp:97 mode=30 >>> >> a=rtpmap:18 G729/8000 >>> >> a=fmtp:18 annexb=no >>> >> a=rtpmap:112 G726-32/8000 >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-16 >>> >> a=silenceSupp:off - - - - >>> >> a=ptime:20 >>> >> >>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>> sofia_glue_negotiate_sdp() >>> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] >>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>> sofia_glue_negotiate_sdp() >>> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] >>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>> sofia_glue_negotiate_sdp() >>> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>> sofia_glue_negotiate_sdp() >>> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 >>> sofia_glue_tech_set_codec() >>> >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 >>> >> samples >>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 >>> sofia_glue_negotiate_sdp() >>> >> Set 2833 dtmf payload to 101 >>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() >>> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> >>> CS_INIT >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>> >> 07771236762 at sipgate.co.uk [BREAK] >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> >> Running State Change CS_INIT >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> INIT >>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() >>> sofia/external/ >>> >> 07771236762 at sipgate.co.uk SOFIA INIT >>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() >>> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> >>> >> CS_ROUTING >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>> >> 07771236762 at sipgate.co.uk [BREAK] >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> INIT going to sleep >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> >> Running State Change CS_ROUTING >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> ROUTING >>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() >>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING >>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 >>> >> switch_core_standard_on_routing() >>> >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING >>> >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>> >> Processing 07771236762->00442083324655 in context public >>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing >>> >> [public->skype_uri] continue=false >>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) >>> >> [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ >>> >> break=on-false >>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action >>> >> bridge(skypiax/skypiax1/xyzTestUK) >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 >>> >> switch_core_standard_on_routing() (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>> >> 07771236762 at sipgate.co.uk [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> ROUTING going to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> >> Running State Change CS_EXECUTE >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> EXECUTE >>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() >>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 >>> >> switch_core_standard_on_execute() >>> >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE >>> >> EXECUTE >>> >> >>> sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 >>> channel_outgoing_channel() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?585 ?][ ? ? ? ? ?][-1, 0, 0] >>> >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?151 ?][skypiax1 ?][-1, 0, 0] >>> >> skypiax_codec >>> >> SUCCESS >>> >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 >>> >> switch_channel_set_name() >>> >> New Channel skypiax/skypiax1/xyzTestUK >>> >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?773 ?][skypiax1 ?][-1, 0, 0] Calling >>> >> Skype, rdest is: xyzTestUK >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AGC OFF|||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AEC OFF|||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 >>> channel_outgoing_channel() >>> >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal >>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>> >> Change >>> >> CS_INIT >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() >>> >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal >>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?182 ?][skypiax1 ?][-1, 0, 0] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL INIT >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going >>> to >>> >> sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>> >> Change >>> >> CS_ROUTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?257 ?][skypiax1 ?][-1, 0, 0] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 >>> >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change >>> >> CS_ROUTING >>> >> -> CS_CONSUME_MEDIA >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal >>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >>> going >>> >> to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>> >> Change >>> >> CS_CONSUME_MEDIA >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State >>> CONSUME_MEDIA >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||AGC OFF||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||AEC OFF||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>> >> READING: >>> >> |||CALL 455 STATUS UNPLACED||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 0, 0] >>> Skype >>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >>> >> UNPLACED,where: >>> >> NULL! >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?371 ?][skypiax1 ?][-1, 3,116] >>> >> skype_call: 455 is now UNPLACED >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,116] >>> >> READING: >>> >> |||CALL 455 STATUS ROUTING||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,116] >>> Skype >>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >>> >> ROUTING,where: >>> >> NULL! >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?365 ?][skypiax1 ?][-1, 3,117] >>> >> skype_call: 455 is now ROUTING >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >>> >> READING: >>> >> |||CALL 455 FAILUREREASON 7||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >>> Skype >>> >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: >>> >> 7,where: >>> >> NULL! >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?201 ?][skypiax1 ?][-1, 3,117] >>> Skype >>> >> FAILED on skype_call 455. Let's wait for the FAILED message. >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >>> >> READING: >>> >> |||CALL 455 VAA_INPUT_STATUS FALSE||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >>> Skype >>> >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: >>> >> FALSE,where: NULL! >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >>> >> READING: >>> >> |||CALL 455 STATUS FAILED||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >>> Skype >>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >>> FAILED,where: >>> >> NULL! >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 >>> >> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?334 ?][skypiax1 ?][-1, 3,112] we >>> >> tried >>> >> to call Skype on skype_call 455 and Skype has now FAILED >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> 672 >>> >> ?][skypiax1 ?][-1, 1,112] skype call ended >>> >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 >>> >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK >>> >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >>> >> switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK >>> >> [KILL] >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?293 ?][skypiax1 ?][-1, 1,112] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal >>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 1,112] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 >>> >> switch_ivr_originate() Originate Resulted in Error Cause: 16 >>> >> [NORMAL_CLEARING] >>> >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() >>> >> Originate Failed. ?Cause: NORMAL_CLEARING >>> >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() >>> >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] >>> >> [NORMAL_CLEARING] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >>> >> switch_channel_perform_hangup() Send signal sofia/external/ >>> >> 07771236762 at sipgate.co.uk [KILL] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>> >> 07771236762 at sipgate.co.uk [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> EXECUTE going to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> >> Running State Change CS_HANGUP >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> HANGUP >>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >>> >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: >>> >> NORMAL_CLEARING >>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >>> Responding >>> >> to >>> >> INVITE with: 480 >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >>> >> switch_core_standard_on_hangup() >>> >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: >>> >> NORMAL_CLEARING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> HANGUP going to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> Change CS_HANGUP -> CS_REPORTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>> >> 07771236762 at sipgate.co.uk [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> >> Running State Change CS_REPORTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >>> >> switch_core_session_reporting_state() (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) State REPORTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State >>> CONSUME_MEDIA >>> >> going to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>> >> Change >>> >> CS_HANGUP >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?228 ?][skypiax1 ?][-1, 1,112] hanging >>> up >>> >> skype call: 455 >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>> >> ][skypiax1 ?][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?235 ?][skypiax1 ?][-1, 1,112] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >>> >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard >>> >> HANGUP, >>> >> cause: NORMAL_CLEARING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >>> going >>> >> to >>> >> sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >>> >> CS_HANGUP -> CS_REPORTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>> >> switch_core_session_signal_state_change() Send signal >>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>> rev >>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 1,112] >>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>> >> Change >>> >> CS_REPORTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >>> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) >>> State >>> >> REPORTING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 >>> >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard >>> >> REPORTING, cause: NORMAL_CLEARING >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >>> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) >>> State >>> >> REPORTING going to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >>> >> CS_REPORTING -> CS_DESTROY >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 >>> >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >>> >> Locked, >>> >> Waiting on external entities >>> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 >>> >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >>> Ended >>> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 >>> >> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK >>> >> [CS_DESTROY] >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >>> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >>> >> DESTROY >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 >>> >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard >>> >> DESTROY >>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >>> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >>> >> DESTROY going to sleep >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 1,112] >>> >> READING: >>> >> |||ERROR 559 CALL: Action failed||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?91 ? ][skypiax1 ?][-1, 1,112] >>> Skype >>> >> got ERROR: |||ERROR||| >>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 >>> skypiax_signaling_read() >>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?93 ? ][skypiax1 ?][-1, 1,110] >>> >> skype_call now is DOWN >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> 672 >>> >> ?][skypiax1 ?][-1, 1,110] skype call ended >>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 >>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>> 687 >>> >> ?][skypiax1 ?][-1, 1,110] no session >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 >>> >> switch_core_standard_on_reporting() sofia/external/ >>> >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 >>> >> switch_core_session_reporting_state() (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 >>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>> State >>> >> Change CS_REPORTING -> CS_DESTROY >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 >>> >> switch_core_session_thread() Session 1 (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities >>> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 >>> >> switch_core_session_thread() Session 1 (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) Ended >>> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 >>> >> switch_core_session_thread() Close Channel sofia/external/ >>> >> 07771236762 at sipgate.co.uk [CS_DESTROY] >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >>> >> switch_core_session_destroy_state() (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) State DESTROY >>> >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 >>> >> switch_core_standard_on_destroy() >>> >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY >>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >>> >> switch_core_session_destroy_state() (sofia/external/ >>> >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep >>> >> -- >>> >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + >>> >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* >>> >> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at 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 at hotmail.com >>> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> > iax:guest at conference.freeswitch.org/888 >>> > >>> googletalk:conf+888 at conference.freeswitch.org >>> > pstn:213-799-1400 >>> > >>> >>> -- >>> Sent from my mobile device >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> ========================================= >>> www.celliax.org >>> via Pierlombardo 9, 20135 Milano >>> Italy >>> gmaruzz at celliax dot org >>> Cell : +39-347-2665618 >>> Fax : +39-02-87390039 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From mszlazak at aol.com Fri May 1 12:47:40 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 01 May 2009 15:47:40 -0400 Subject: [Freeswitch-users] Latest SVN update gives Windows Express compiler errors ... In-Reply-To: <5BFA959A-5517-43BC-BA22-205791AD659B@jerris.com> References: <8CB98298639AEA6-280-33C3@webmail-dx08.sysops.aol.com> <5BFA959A-5517-43BC-BA22-205791AD659B@jerris.com> Message-ID: <8CB98ACD216FAE2-E0C-4FA@WEBMAIL-DC11.sysops.aol.com> I updated again earlier today and they're gone. -----Original Message----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, 1 May 2009 7:43 am Subject: Re: [Freeswitch-users] Latest SVN update gives Windows Express compiler errors ... Do you have any?specifics?of the errors? Mike On May 1, 2009, at 12:07 AM, mszlazak at aol.com wrote: I'm getting Windows Express compiler errors on the latest svn update to trunk 13213. It looks like the path is wrong to some files. Instead of folder "Debug", it's looking for files in folder "Debug DLL" Mark. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/8dd79141/attachment.html From gmaruzz at celliax.org Fri May 1 13:26:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 1 May 2009 22:26:44 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <7b197bef0905011225t525dc47cu2c3f8c9b548e4600@mail.gmail.com> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> <20090501111945.168380@gmx.net> <7b197bef0905010714l4fc38792o63877627704c1939@mail.gmail.com> <7b197bef0905011225t525dc47cu2c3f8c9b548e4600@mail.gmail.com> Message-ID: <7b197bef0905011326u5d158155h671d7de38b540038@mail.gmail.com> Hi Phil, I had to close the Jira, please try again with your original alsa.conf. Your editing of it was probably causing some of the new problems. I just tested it all in a virtual machine (using virtualbox) and it worked for me. Only things that comes at my mind is that I used the 32bit, not the 64bit version. You are using 64bit in a Xen environment (if I understood correctly), but others have done it with success (btw, various deployment in Amazon ec2). The error you was receiving in the original post (ERROR 7) is the Skype client not finding the sound device. Maybe is just a problem of permissions? The user the Skype client instance is started as has permission to read/write on the sound device? Have you tried it starting Skype instance as root user? In my test deployment here, ls -l /dev/snd/* shows that the devices are r/w only by root... Change the permission of the devices if you start Skype as another user. chmod -R a+rw /dev/snd So, please go back to the original alsa.conf ( I will mail it to your address), then be sure to follow all the steps. Then, as a first test, try a call to "echo123" that is the test call answering machine made available by Skype. Let me know. -giovanni On Fri, May 1, 2009 at 9:25 PM, Giovanni Maruzzelli wrote: > Hi Phil, > > I just tried all the steps (exactly, just cut and paste) from the wiki page: > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS_5.2.2C_from_scratch > > I substituted 5.3 instead of 5.2. > > I'm afraid it worked flawlessly for me. (shocked about: Anthony is > right about CentOS being "boring and predictable", good qualities for > a server OS!) > > At the start of Skype clients it will tell bizarre things about hdmi, > but they are unharmful (I've not edited the alsa stuff, it still groak > about non-existent hdmi, but it works nonetheless). > > So, I suspect your problems have some other cause. > > Now I go read the Jira and the attached files, and I hope to be more of help. > > -giovanni > > > On Fri, May 1, 2009 at 4:14 PM, Giovanni Maruzzelli wrote: >> Gruss Phil, >> >> actually it was shooting in the dark from my side, because I not yet >> tested centos5.3, only centos5.2 >> >> As soon as I test it out I'll be back to you. >> Thanks for filing the Jira. >> >> -giovanni >> >> >> On Fri, May 1, 2009 at 1:19 PM, ? wrote: >>> Ciao Giovanni, >>> >>> grazie per la tua risposta. Removing 'hdmi' did make some changes, but it >>> still doesn't work. I have filed a jira: >>> >>> http://jira.freeswitch.org/browse/MODSKYPIAX-33 >>> >>> Buon primo maggio anche a te, >>> Phil >>> >>> -------- Original-Nachricht -------- >>>> Datum: Fri, 1 May 2009 08:20:10 +0200 >>>> Von: Giovanni Maruzzelli >>>> An: freeswitch-users at lists.freeswitch.org >>>> Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O ? ?error >>> >>>> Have a happy MayDay! >>>> >>>> I cannot see the whole mail now, it's clipped for my mobile, but it >>>> seems the nth bizarry of new alsa config file, that creates an hdmi >>>> device even if you do not have one. Try to edit >>>> /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and >>>> delete any mention of 'hdmi'. >>>> If this do not works, please file a jira or write again. >>>> Giovanni >>>> >>>> >>>> >>>> On 5/1/09, Anthony Minessale wrote: >>>> > if you put that info in a jira ticket >>>> > >>>> > http://jira.freeswitch.org >>>> > >>>> > and route it to skypeiax , the guy who maintains that module will see >>>> it. >>>> > >>>> > >>>> > On Thu, Apr 30, 2009 at 5:37 PM, wrote: >>>> > >>>> >> >>>> >> Hello, >>>> >> >>>> >> I am trying to get skypiax working, but I am having trouble with the >>>> >> sound. >>>> >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and >>>> >> I am getting the following error: >>>> >> >>>> >> ? ? ? ?ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM >>>> >> cards.pcm.hdmi >>>> >> >>>> >> >>>> >> I am running centos 5.3 and have followed the installation guide on the >>>> >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When >>>> >> saving >>>> >> the configuration on my desktop I have set the sound card to snd_dummy. >>>> On >>>> >> the server the startup script load snd-dumy like this /sbin/modprobe >>>> >> snd-dummy enable=1. >>>> >> Below is the output of lsmod and the debug output from FS. It would be >>>> >> great if someone could help me fix my problem. >>>> >> >>>> >> Thank you very much. >>>> >> Best wishes, >>>> >> Phil >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> -bash-3.2# lsmod >>>> >> Module ? ? ? ? ? ? ? ? ?Size ?Used by >>>> >> snd_dummy ? ? ? ? ? ? ?12416 ?0 >>>> >> snd_seq_oss ? ? ? ? ? ?32832 ?0 >>>> >> snd_seq_midi_event ? ? ?7744 ?1 snd_seq_oss >>>> >> snd_seq ? ? ? ? ? ? ? ?55200 ?4 snd_seq_oss,snd_seq_midi_event >>>> >> snd_seq_device ? ? ? ? ?7120 ?1 snd_seq_oss >>>> >> snd_pcm_oss ? ? ? ? ? ?44480 ?0 >>>> >> snd_mixer_oss ? ? ? ? ?16512 ?1 snd_pcm_oss >>>> >> snd_pcm ? ? ? ? ? ? ? ?79624 ?2 snd_dummy,snd_pcm_oss >>>> >> snd_timer ? ? ? ? ? ? ?22088 ?2 snd_seq,snd_pcm >>>> >> snd ? ? ? ? ? ? ? ? ? ?55976 ?8 >>>> >> >>>> snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer >>>> >> soundcore ? ? ? ? ? ? ? 7456 ?1 snd >>>> >> snd_page_alloc ? ? ? ? ?8720 ?1 snd_pcm >>>> >> >>>> >> >>>> >> >>>> >> freeswitch at voipserverServerFreeswitch> load mod_skypiax >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?718 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.debug=0 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?720 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.debug=8 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?731 ?][none ? ? ?][-1,-1,-1] >>>> >> codec-master >>>> >> globals.debug=8 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?734 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.dialplan=XML >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?740 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.context=default >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?743 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.codec_string=gsm,ulaw >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?750 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.codec_rates_string=8000,16000 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?723 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.hold_music= >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?737 ?][none ? ? ?][-1,-1,-1] >>>> >> globals.destination=5000 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? ?][-1,-1,-1] >>>> >> interface_id=1 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? ?][-1,-1,-1] >>>> >> name=skypiax1 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? ?][-1,-1,-1] >>>> Initialized >>>> >> XInitThreads! >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax1 ?][-1, 0, 0] >>>> CONFIGURING >>>> >> interface_id=1 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 >>>> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 >>>> globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 >>>> globals.SKYPIAX_INTERFACES[interface_id].destination=3101 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax1 ?][-1, 0, 0] >>>> >> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default >>>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax1 ?][-1, 0, 0] STARTING >>>> >> interface_id=1 >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >>>> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> >> 1407 >>>> >> ][skypiax1 ?][-1, 0, 0] X Display ':101' opened >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? ?][-1,-1,-1] Skype >>>> >> instance found with id #2097454 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> 661 >>>> >> ?][skypiax1 ?][-1, 0, 0] In skypiax_signaling_thread_func: started, >>>> >> p=0x2aaab93226f8 >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||OK||| >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||PROTOCOL 7||| >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||CONNSTATUS ONLINE||| >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||CURRENTUSERHANDLE xyzUK||| >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax1 ?][-1, 0, 0] >>>> Skype >>>> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >>>> >> cuh: >>>> >> xyzUK, skype_user: xyzUK! >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||USERSTATUS ONLINE||| >>>> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax1 ?][-1, 0, 0] WAITING >>>> roughly >>>> >> 10 >>>> >> seconds to find a running Skype client and connect to its SKYPE API for >>>> >> interface_id=1 >>>> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax1 ?][-1, 0, 0] Found a >>>> running >>>> >> Skype client, connected to its SKYPE API for interface_id=1, waiting 60 >>>> >> seconds for CURRENTUSERHANDLE==xyzUK >>>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax1 ?][-1, 0, 0] >>>> Interface_id=1 >>>> >> is now STARTED, the Skype client to which we are connected gave us the >>>> >> correct CURRENTUSERHANDLE (xyzUK) >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? ?][-1,-1,-1] >>>> >> interface_id=2 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? ?][-1,-1,-1] >>>> >> name=skypiax2 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? ?][-1,-1,-1] >>>> Initialized >>>> >> XInitThreads! >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax2 ?][-1, 0, 0] >>>> CONFIGURING >>>> >> interface_id=2 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 >>>> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 >>>> >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 >>>> globals.SKYPIAX_INTERFACES[interface_id].destination=5000 >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax2 ?][-1, 0, 0] >>>> >> interface_id=2 globals.SKYPIAX_INTERFACES[interface_id].context=default >>>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax2 ?][-1, 0, 0] STARTING >>>> >> interface_id=2 >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 >>>> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> >> 1407 >>>> >> ][skypiax2 ?][-1, 0, 0] X Display ':102' opened >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 skypiax_present() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? ?][-1,-1,-1] Skype >>>> >> instance found with id #2097454 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> 661 >>>> >> ?][skypiax2 ?][-1, 0, 0] In skypiax_signaling_thread_func: started, >>>> >> p=0x2aaab9325c18 >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||OK||| >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||PROTOCOL 7||| >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||CONNSTATUS ONLINE||| >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||CURRENTUSERHANDLE voipserver||| >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax2 ?][-1, 0, 0] >>>> Skype >>>> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: CURRENTUSERHANDLE, >>>> >> cuh: >>>> >> voipserver, skype_user: voipserver! >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||USERSTATUS ONLINE||| >>>> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax2 ?][-1, 0, 0] WAITING >>>> roughly >>>> >> 10 >>>> >> seconds to find a running Skype client and connect to its SKYPE API for >>>> >> interface_id=2 >>>> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax2 ?][-1, 0, 0] Found a >>>> running >>>> >> Skype client, connected to its SKYPE API for interface_id=2, waiting 60 >>>> >> seconds for CURRENTUSERHANDLE==voipserver >>>> >> API CALL [load(mod_skypiax)] output: >>>> >> +OK >>>> >> >>>> >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax2 ?][-1, 0, 0] >>>> Interface_id=2 >>>> >> is now STARTED, the Skype client to which we are connected gave us the >>>> >> correct CURRENTUSERHANDLE (voipserver) >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].interface_id=1 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].context=default >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].dialplan=XML >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].destination=3101 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax1 ?][-1, 0, 0] i=1 >>>> >> globals.SKYPIAX_INTERFACES[1].context=default >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].interface_id=2 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].context=default >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].dialplan=XML >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].destination=5000 >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax2 ?][-1, 0, 0] i=2 >>>> >> globals.SKYPIAX_INTERFACES[2].context=default >>>> >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 >>>> >> switch_loadable_module_load_file() Successfully Loaded [mod_skypiax] >>>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 >>>> >> switch_loadable_module_process() Adding Endpoint 'skypiax' >>>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >>>> >> switch_loadable_module_process() Adding API Function 'sk' >>>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 >>>> >> switch_loadable_module_process() Adding API Function 'skypiax' >>>> >> freeswitch at voipserverServerFreeswitch> >>>> >> freeswitch at voipserverServerFreeswitch> >>>> >> freeswitch at voipserverServerFreeswitch> >>>> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] >>>> >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 >>>> >> ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] READING: |||USER paolofun6 >>>> >> PHONE_MOBILE +420775216536||| >>>> >> >>>> >> freeswitch at voipserverServerFreeswitch> >>>> >> freeswitch at voipserverServerFreeswitch> >>>> >> freeswitch at voipserverServerFreeswitch> >>>> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 [NOTICE] >>>> >> switch_channel.c:602 switch_channel_set_name() New Channel >>>> sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() >>>> >> Channel >>>> >> sofia/external/07771236762 at sipgate.co.uk entering state [received][100] >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() >>>> Remote >>>> >> SDP: >>>> >> v=0 >>>> >> o=root 15141 15141 IN IP4 217.10.66.71 >>>> >> s=session >>>> >> c=IN IP4 217.10.66.71 >>>> >> t=0 0 >>>> >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 >>>> >> a=rtpmap:8 PCMA/8000 >>>> >> a=rtpmap:0 PCMU/8000 >>>> >> a=rtpmap:3 GSM/8000 >>>> >> a=rtpmap:97 iLBC/8000 >>>> >> a=fmtp:97 mode=30 >>>> >> a=rtpmap:18 G729/8000 >>>> >> a=fmtp:18 annexb=no >>>> >> a=rtpmap:112 G726-32/8000 >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-16 >>>> >> a=silenceSupp:off - - - - >>>> >> a=ptime:20 >>>> >> >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>>> sofia_glue_negotiate_sdp() >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>>> sofia_glue_negotiate_sdp() >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>>> sofia_glue_negotiate_sdp() >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 >>>> sofia_glue_negotiate_sdp() >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 >>>> sofia_glue_tech_set_codec() >>>> >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms 160 >>>> >> samples >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 >>>> sofia_glue_negotiate_sdp() >>>> >> Set 2833 dtmf payload to 101 >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 sofia_handle_sip_i_state() >>>> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> >>>> CS_INIT >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [BREAK] >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> >> Running State Change CS_INIT >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> INIT >>>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() >>>> sofia/external/ >>>> >> 07771236762 at sipgate.co.uk SOFIA INIT >>>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() >>>> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> >>>> >> CS_ROUTING >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [BREAK] >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> INIT going to sleep >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> >> Running State Change CS_ROUTING >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> ROUTING >>>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() >>>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 >>>> >> switch_core_standard_on_routing() >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING >>>> >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>>> >> Processing 07771236762->00442083324655 in context public >>>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing >>>> >> [public->skype_uri] continue=false >>>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) >>>> >> [skype_uri] destination_number(00442083324655) =~ /^(00442083324655)$/ >>>> >> break=on-false >>>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action >>>> >> bridge(skypiax/skypiax1/xyzTestUK) >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 >>>> >> switch_core_standard_on_routing() (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> ROUTING going to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> >> Running State Change CS_EXECUTE >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> EXECUTE >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() >>>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 >>>> >> switch_core_standard_on_execute() >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE >>>> >> EXECUTE >>>> >> >>>> sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 >>>> channel_outgoing_channel() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?585 ?][ ? ? ? ? ?][-1, 0, 0] >>>> >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?151 ?][skypiax1 ?][-1, 0, 0] >>>> >> skypiax_codec >>>> >> SUCCESS >>>> >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 >>>> >> switch_channel_set_name() >>>> >> New Channel skypiax/skypiax1/xyzTestUK >>>> >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?773 ?][skypiax1 ?][-1, 0, 0] Calling >>>> >> Skype, rdest is: xyzTestUK >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AGC OFF|||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AEC OFF|||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 >>>> channel_outgoing_channel() >>>> >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>>> >> Change >>>> >> CS_INIT >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() >>>> >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?182 ?][skypiax1 ?][-1, 0, 0] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL INIT >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT going >>>> to >>>> >> sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>>> >> Change >>>> >> CS_ROUTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?257 ?][skypiax1 ?][-1, 0, 0] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 >>>> >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change >>>> >> CS_ROUTING >>>> >> -> CS_CONSUME_MEDIA >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, 0] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State ROUTING >>>> going >>>> >> to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>>> >> Change >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State >>>> CONSUME_MEDIA >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||AGC OFF||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||AEC OFF||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] >>>> >> READING: >>>> >> |||CALL 455 STATUS UNPLACED||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 0, 0] >>>> Skype >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >>>> >> UNPLACED,where: >>>> >> NULL! >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?371 ?][skypiax1 ?][-1, 3,116] >>>> >> skype_call: 455 is now UNPLACED >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,116] >>>> >> READING: >>>> >> |||CALL 455 STATUS ROUTING||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,116] >>>> Skype >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >>>> >> ROUTING,where: >>>> >> NULL! >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?365 ?][skypiax1 ?][-1, 3,117] >>>> >> skype_call: 455 is now ROUTING >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >>>> >> READING: >>>> >> |||CALL 455 FAILUREREASON 7||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >>>> Skype >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, value: >>>> >> 7,where: >>>> >> NULL! >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?201 ?][skypiax1 ?][-1, 3,117] >>>> Skype >>>> >> FAILED on skype_call 455. Let's wait for the FAILED message. >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >>>> >> READING: >>>> >> |||CALL 455 VAA_INPUT_STATUS FALSE||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >>>> Skype >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, value: >>>> >> FALSE,where: NULL! >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 3,117] >>>> >> READING: >>>> >> |||CALL 455 STATUS FAILED||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, 3,117] >>>> Skype >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: >>>> FAILED,where: >>>> >> NULL! >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 >>>> >> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?334 ?][skypiax1 ?][-1, 3,112] we >>>> >> tried >>>> >> to call Skype on skype_call 455 and Skype has now FAILED >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> 672 >>>> >> ?][skypiax1 ?][-1, 1,112] skype call ended >>>> >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 >>>> >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK >>>> >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >>>> >> switch_channel_perform_hangup() Send signal skypiax/skypiax1/xyzTestUK >>>> >> [KILL] >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 channel_kill_channel() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?293 ?][skypiax1 ?][-1, 1,112] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 1,112] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 >>>> >> switch_ivr_originate() Originate Resulted in Error Cause: 16 >>>> >> [NORMAL_CLEARING] >>>> >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 audio_bridge_function() >>>> >> Originate Failed. ?Cause: NORMAL_CLEARING >>>> >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 audio_bridge_function() >>>> >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] >>>> >> [NORMAL_CLEARING] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 >>>> >> switch_channel_perform_hangup() Send signal sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [KILL] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> EXECUTE going to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> >> Running State Change CS_HANGUP >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> HANGUP >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >>>> >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: >>>> >> NORMAL_CLEARING >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >>>> Responding >>>> >> to >>>> >> INVITE with: 480 >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: >>>> >> NORMAL_CLEARING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> HANGUP going to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> Change CS_HANGUP -> CS_REPORTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> >> Running State Change CS_REPORTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >>>> >> switch_core_session_reporting_state() (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) State REPORTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State >>>> CONSUME_MEDIA >>>> >> going to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>>> >> Change >>>> >> CS_HANGUP >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?228 ?][skypiax1 ?][-1, 1,112] hanging >>>> up >>>> >> skype call: 455 >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1262 >>>> >> ][skypiax1 ?][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?235 ?][skypiax1 ?][-1, 1,112] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard >>>> >> HANGUP, >>>> >> cause: NORMAL_CLEARING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP >>>> going >>>> >> to >>>> >> sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >>>> >> CS_HANGUP -> CS_REPORTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 >>>> >> switch_core_session_signal_state_change() Send signal >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() >>>> rev >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 1,112] >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State >>>> >> Change >>>> >> CS_REPORTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >>>> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) >>>> State >>>> >> REPORTING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 >>>> >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard >>>> >> REPORTING, cause: NORMAL_CLEARING >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 >>>> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) >>>> State >>>> >> REPORTING going to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change >>>> >> CS_REPORTING -> CS_DESTROY >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 >>>> >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >>>> >> Locked, >>>> >> Waiting on external entities >>>> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 >>>> >> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) >>>> Ended >>>> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 >>>> >> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK >>>> >> [CS_DESTROY] >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >>>> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >>>> >> DESTROY >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 >>>> >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard >>>> >> DESTROY >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 >>>> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State >>>> >> DESTROY going to sleep >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 1,112] >>>> >> READING: >>>> >> |||ERROR 559 CALL: Action failed||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?91 ? ][skypiax1 ?][-1, 1,112] >>>> Skype >>>> >> got ERROR: |||ERROR||| >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 >>>> skypiax_signaling_read() >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?93 ? ][skypiax1 ?][-1, 1,110] >>>> >> skype_call now is DOWN >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> 672 >>>> >> ?][skypiax1 ?][-1, 1,110] skype call ended >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE >>>> 687 >>>> >> ?][skypiax1 ?][-1, 1,110] no session >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 >>>> >> switch_core_standard_on_reporting() sofia/external/ >>>> >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 >>>> >> switch_core_session_reporting_state() (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (sofia/external/07771236762 at sipgate.co.uk) >>>> State >>>> >> Change CS_REPORTING -> CS_DESTROY >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 >>>> >> switch_core_session_thread() Session 1 (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities >>>> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 >>>> >> switch_core_session_thread() Session 1 (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) Ended >>>> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 >>>> >> switch_core_session_thread() Close Channel sofia/external/ >>>> >> 07771236762 at sipgate.co.uk [CS_DESTROY] >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >>>> >> switch_core_session_destroy_state() (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) State DESTROY >>>> >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >>>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 >>>> >> switch_core_standard_on_destroy() >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 >>>> >> switch_core_session_destroy_state() (sofia/external/ >>>> >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep >>>> >> -- >>>> >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + >>>> >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* >>>> >> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at 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 at hotmail.com >>>> > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> > IRC: irc.freenode.net #freeswitch >>>> > >>>> > FreeSWITCH Developer Conference >>>> > sip:888 at conference.freeswitch.org >>>> > iax:guest at conference.freeswitch.org/888 >>>> > >>>> googletalk:conf+888 at conference.freeswitch.org >>>> > pstn:213-799-1400 >>>> > >>>> >>>> -- >>>> Sent from my mobile device >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> ========================================= >>>> www.celliax.org >>>> via Pierlombardo 9, 20135 Milano >>>> Italy >>>> gmaruzz at celliax dot org >>>> Cell : +39-347-2665618 >>>> Fax : +39-02-87390039 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> -- >>> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > From can_man at gmx.de Fri May 1 15:47:49 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Sat, 02 May 2009 00:47:49 +0200 Subject: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error In-Reply-To: <7b197bef0905011326u5d158155h671d7de38b540038@mail.gmail.com> References: <20090430223701.280500@gmx.net> <191c3a030904301602i7f37c8e2uefe3c73c956bc4@mail.gmail.com> <7b197bef0904302320t6d025985vc4e912b4373577b1@mail.gmail.com> <20090501111945.168380@gmx.net> <7b197bef0905010714l4fc38792o63877627704c1939@mail.gmail.com> <7b197bef0905011225t525dc47cu2c3f8c9b548e4600@mail.gmail.com> <7b197bef0905011326u5d158155h671d7de38b540038@mail.gmail.com> Message-ID: <20090501224749.170580@gmx.net> Hello, thank you very much for your help. So annoying it was actually a permissions problem, I had checked but missed the one that was crutial. Cheers, Phil -------- Original-Nachricht -------- > Datum: Fri, 1 May 2009 22:26:44 +0200 > Von: Giovanni Maruzzelli > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error > Hi Phil, > > I had to close the Jira, please try again with your original > alsa.conf. Your editing of it was probably causing some of the new > problems. > > I just tested it all in a virtual machine (using virtualbox) and it > worked for me. > > Only things that comes at my mind is that I used the 32bit, not the > 64bit version. > > You are using 64bit in a Xen environment (if I understood correctly), > but others have done it with success (btw, various deployment in > Amazon ec2). > > The error you was receiving in the original post (ERROR 7) is the > Skype client not finding the sound device. > > Maybe is just a problem of permissions? The user the Skype client > instance is started as has permission to read/write on the sound > device? > > Have you tried it starting Skype instance as root user? > > In my test deployment here, ls -l /dev/snd/* shows that the devices > are r/w only by root... > Change the permission of the devices if you start Skype as another user. > > chmod -R a+rw /dev/snd > > So, please go back to the original alsa.conf ( I will mail it to your > address), then be sure to follow all the steps. > > Then, as a first test, try a call to "echo123" that is the test call > answering machine made available by Skype. > > Let me know. > > -giovanni > > > On Fri, May 1, 2009 at 9:25 PM, Giovanni Maruzzelli > wrote: > > Hi Phil, > > > > I just tried all the steps (exactly, just cut and paste) from the wiki > page: > > > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS_5.2.2C_from_scratch > > > > I substituted 5.3 instead of 5.2. > > > > I'm afraid it worked flawlessly for me. (shocked about: Anthony is > > right about CentOS being "boring and predictable", good qualities for > > a server OS!) > > > > At the start of Skype clients it will tell bizarre things about hdmi, > > but they are unharmful (I've not edited the alsa stuff, it still groak > > about non-existent hdmi, but it works nonetheless). > > > > So, I suspect your problems have some other cause. > > > > Now I go read the Jira and the attached files, and I hope to be more of > help. > > > > -giovanni > > > > > > On Fri, May 1, 2009 at 4:14 PM, Giovanni Maruzzelli > wrote: > >> Gruss Phil, > >> > >> actually it was shooting in the dark from my side, because I not yet > >> tested centos5.3, only centos5.2 > >> > >> As soon as I test it out I'll be back to you. > >> Thanks for filing the Jira. > >> > >> -giovanni > >> > >> > >> On Fri, May 1, 2009 at 1:19 PM, ? wrote: > >>> Ciao Giovanni, > >>> > >>> grazie per la tua risposta. Removing 'hdmi' did make some changes, but > it > >>> still doesn't work. I have filed a jira: > >>> > >>> http://jira.freeswitch.org/browse/MODSKYPIAX-33 > >>> > >>> Buon primo maggio anche a te, > >>> Phil > >>> > >>> -------- Original-Nachricht -------- > >>>> Datum: Fri, 1 May 2009 08:20:10 +0200 > >>>> Von: Giovanni Maruzzelli > >>>> An: freeswitch-users at lists.freeswitch.org > >>>> Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = > Sound I/O ? ?error > >>> > >>>> Have a happy MayDay! > >>>> > >>>> I cannot see the whole mail now, it's clipped for my mobile, but it > >>>> seems the nth bizarry of new alsa config file, that creates an hdmi > >>>> device even if you do not have one. Try to edit > >>>> /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir > and > >>>> delete any mention of 'hdmi'. > >>>> If this do not works, please file a jira or write again. > >>>> Giovanni > >>>> > >>>> > >>>> > >>>> On 5/1/09, Anthony Minessale wrote: > >>>> > if you put that info in a jira ticket > >>>> > > >>>> > http://jira.freeswitch.org > >>>> > > >>>> > and route it to skypeiax , the guy who maintains that module will > see > >>>> it. > >>>> > > >>>> > > >>>> > On Thu, Apr 30, 2009 at 5:37 PM, wrote: > >>>> > > >>>> >> > >>>> >> Hello, > >>>> >> > >>>> >> I am trying to get skypiax working, but I am having trouble with > the > >>>> >> sound. > >>>> >> The calls fail with CALL FAILUREREASON 7 = Sound I/O error and > >>>> >> I am getting the following error: > >>>> >> > >>>> >> ? ? ? ?ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > >>>> >> cards.pcm.hdmi > >>>> >> > >>>> >> > >>>> >> I am running centos 5.3 and have followed the installation guide > on the > >>>> >> wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. > When > >>>> >> saving > >>>> >> the configuration on my desktop I have set the sound card to > snd_dummy. > >>>> On > >>>> >> the server the startup script load snd-dumy like this > /sbin/modprobe > >>>> >> snd-dummy enable=1. > >>>> >> Below is the output of lsmod and the debug output from FS. It > would be > >>>> >> great if someone could help me fix my problem. > >>>> >> > >>>> >> Thank you very much. > >>>> >> Best wishes, > >>>> >> Phil > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> -bash-3.2# lsmod > >>>> >> Module ? ? ? ? ? ? ? ? ?Size ?Used by > >>>> >> snd_dummy ? ? ? ? ? ? ?12416 ?0 > >>>> >> snd_seq_oss ? ? ? ? ? ?32832 ?0 > >>>> >> snd_seq_midi_event ? ? ?7744 ?1 snd_seq_oss > >>>> >> snd_seq ? ? ? ? ? ? ? ?55200 ?4 > snd_seq_oss,snd_seq_midi_event > >>>> >> snd_seq_device ? ? ? ? ?7120 ?1 snd_seq_oss > >>>> >> snd_pcm_oss ? ? ? ? ? ?44480 ?0 > >>>> >> snd_mixer_oss ? ? ? ? ?16512 ?1 snd_pcm_oss > >>>> >> snd_pcm ? ? ? ? ? ? ? ?79624 ?2 snd_dummy,snd_pcm_oss > >>>> >> snd_timer ? ? ? ? ? ? ?22088 ?2 snd_seq,snd_pcm > >>>> >> snd ? ? ? ? ? ? ? ? ? ?55976 ?8 > >>>> >> > >>>> > snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > >>>> >> soundcore ? ? ? ? ? ? ? 7456 ?1 snd > >>>> >> snd_page_alloc ? ? ? ? ?8720 ?1 snd_pcm > >>>> >> > >>>> >> > >>>> >> > >>>> >> freeswitch at voipserverServerFreeswitch> load mod_skypiax > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?718 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.debug=0 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?720 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.debug=8 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?731 ?][none ? ? > ?][-1,-1,-1] > >>>> >> codec-master > >>>> >> globals.debug=8 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?734 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.dialplan=XML > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?740 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.context=default > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?743 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.codec_string=gsm,ulaw > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?750 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.codec_rates_string=8000,16000 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?723 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.hold_music= > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?737 ?][none ? ? > ?][-1,-1,-1] > >>>> >> globals.destination=5000 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? > ?][-1,-1,-1] > >>>> >> interface_id=1 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? > ?][-1,-1,-1] > >>>> >> name=skypiax1 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? > ?][-1,-1,-1] > >>>> Initialized > >>>> >> XInitThreads! > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax1 ?][-1, 0, > 0] > >>>> CONFIGURING > >>>> >> interface_id=1 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > >>>> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > >>>> globals.SKYPIAX_INTERFACES[interface_id].skype_user=xyzUK > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].name=skypiax1 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].context=default > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > >>>> globals.SKYPIAX_INTERFACES[interface_id].destination=3101 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> interface_id=1 > globals.SKYPIAX_INTERFACES[interface_id].context=default > >>>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax1 ?][-1, 0, 0] > STARTING > >>>> >> interface_id=1 > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > >>>> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> >> 1407 > >>>> >> ][skypiax1 ?][-1, 0, 0] X Display ':101' opened > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 > skypiax_present() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? > ?][-1,-1,-1] Skype > >>>> >> instance found with id #2097454 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:661 > >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> 661 > >>>> >> ?][skypiax1 ?][-1, 0, 0] In skypiax_signaling_thread_func: > started, > >>>> >> p=0x2aaab93226f8 > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||OK||| > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||PROTOCOL 7||| > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||CONNSTATUS ONLINE||| > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||CURRENTUSERHANDLE xyzUK||| > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:111 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax1 ?][-1, > 0, 0] > >>>> Skype > >>>> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: > CURRENTUSERHANDLE, > >>>> >> cuh: > >>>> >> xyzUK, skype_user: xyzUK! > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||USERSTATUS ONLINE||| > >>>> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:976 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax1 ?][-1, 0, 0] > WAITING > >>>> roughly > >>>> >> 10 > >>>> >> seconds to find a running Skype client and connect to its SKYPE > API for > >>>> >> interface_id=1 > >>>> >> 2009-04-30 17:47:35 [NOTICE] mod_skypiax.c:986 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax1 ?][-1, 0, 0] > Found a > >>>> running > >>>> >> Skype client, connected to its SKYPE API for interface_id=1, > waiting 60 > >>>> >> seconds for CURRENTUSERHANDLE==xyzUK > >>>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:1004 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax1 ?][-1, 0, 0] > >>>> Interface_id=1 > >>>> >> is now STARTED, the Skype client to which we are connected gave us > the > >>>> >> correct CURRENTUSERHANDLE (xyzUK) > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?847 ?][none ? ? > ?][-1,-1,-1] > >>>> >> interface_id=2 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?870 ?][none ? ? > ?][-1,-1,-1] > >>>> >> name=skypiax2 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?876 ?][none ? ? > ?][-1,-1,-1] > >>>> Initialized > >>>> >> XInitThreads! > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?897 ?][skypiax2 ?][-1, 0, > 0] > >>>> CONFIGURING > >>>> >> interface_id=2 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:920 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?920 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > >>>> globals.SKYPIAX_INTERFACES[interface_id].X11_display=:102 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:924 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?924 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > >>>> >> globals.SKYPIAX_INTERFACES[interface_id].skype_user=voipserver > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:928 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?928 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15558 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:932 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?932 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > >>>> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15559 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:935 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?935 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].name=skypiax2 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:938 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?938 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].context=default > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:942 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?942 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:946 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?946 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > >>>> globals.SKYPIAX_INTERFACES[interface_id].destination=5000 > >>>> >> 2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:949 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?949 ?][skypiax2 ?][-1, 0, > 0] > >>>> >> interface_id=2 > globals.SKYPIAX_INTERFACES[interface_id].context=default > >>>> >> 2009-04-30 17:47:35 [WARNING] mod_skypiax.c:950 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?950 ?][skypiax2 ?][-1, 0, 0] > STARTING > >>>> >> interface_id=2 > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1407 > >>>> >> skypiax_do_skypeapi_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> >> 1407 > >>>> >> ][skypiax2 ?][-1, 0, 0] X Display ':102' opened > >>>> >> 2009-04-30 17:47:35 [DEBUG] skypiax_protocol.c:1309 > skypiax_present() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1309 ][none ? ? > ?][-1,-1,-1] Skype > >>>> >> instance found with id #2097454 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:661 > >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> 661 > >>>> >> ?][skypiax2 ?][-1, 0, 0] In skypiax_signaling_thread_func: > started, > >>>> >> p=0x2aaab9325c18 > >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||OK||| > >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||PROTOCOL 7||| > >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||CONNSTATUS ONLINE||| > >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||CURRENTUSERHANDLE voipserver||| > >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:111 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?111 ?][skypiax2 ?][-1, > 0, 0] > >>>> Skype > >>>> >> MSG: message: CURRENTUSERHANDLE, currentuserhandle: > CURRENTUSERHANDLE, > >>>> >> cuh: > >>>> >> voipserver, skype_user: voipserver! > >>>> >> 2009-04-30 17:47:36 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax2 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||USERSTATUS ONLINE||| > >>>> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:976 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?976 ?][skypiax2 ?][-1, 0, 0] > WAITING > >>>> roughly > >>>> >> 10 > >>>> >> seconds to find a running Skype client and connect to its SKYPE > API for > >>>> >> interface_id=2 > >>>> >> 2009-04-30 17:47:36 [NOTICE] mod_skypiax.c:986 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][NOTICA ?986 ?][skypiax2 ?][-1, 0, 0] > Found a > >>>> running > >>>> >> Skype client, connected to its SKYPE API for interface_id=2, > waiting 60 > >>>> >> seconds for CURRENTUSERHANDLE==voipserver > >>>> >> API CALL [load(mod_skypiax)] output: > >>>> >> +OK > >>>> >> > >>>> >> 2009-04-30 17:47:36 [WARNING] mod_skypiax.c:1004 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][WARNINGA ?1004 ][skypiax2 ?][-1, 0, 0] > >>>> Interface_id=2 > >>>> >> is now STARTED, the Skype client to which we are connected gave us > the > >>>> >> correct CURRENTUSERHANDLE (voipserver) > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].interface_id=1 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].X11_display=:101 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].name=skypiax1 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].context=default > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].dialplan=XML > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].destination=3101 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax1 ?][-1, 0, 0] > i=1 > >>>> >> globals.SKYPIAX_INTERFACES[1].context=default > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1028 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1028 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].interface_id=2 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1030 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1030 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].X11_display=:102 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1032 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1032 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].name=skypiax2 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1034 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1034 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].context=default > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1036 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1036 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].dialplan=XML > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1038 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1038 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].destination=5000 > >>>> >> 2009-04-30 17:47:36 [DEBUG] mod_skypiax.c:1040 load_config() rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?1040 ][skypiax2 ?][-1, 0, 0] > i=2 > >>>> >> globals.SKYPIAX_INTERFACES[2].context=default > >>>> >> 2009-04-30 17:47:36 [CONSOLE] switch_loadable_module.c:889 > >>>> >> switch_loadable_module_load_file() Successfully Loaded > [mod_skypiax] > >>>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:142 > >>>> >> switch_loadable_module_process() Adding Endpoint 'skypiax' > >>>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > >>>> >> switch_loadable_module_process() Adding API Function 'sk' > >>>> >> 2009-04-30 17:47:36 [NOTICE] switch_loadable_module.c:270 > >>>> >> switch_loadable_module_process() Adding API Function 'skypiax' > >>>> >> freeswitch at voipserverServerFreeswitch> > >>>> >> freeswitch at voipserverServerFreeswitch> > >>>> >> freeswitch at voipserverServerFreeswitch> > >>>> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:41 [DEBUG] > >>>> >> skypiax_protocol.c:57 skypiax_signaling_read() rev 13177[(nil)|37 > >>>> >> ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, 0, 0] READING: |||USER > paolofun6 > >>>> >> PHONE_MOBILE +420775216536||| > >>>> >> > >>>> >> freeswitch at voipserverServerFreeswitch> > >>>> >> freeswitch at voipserverServerFreeswitch> > >>>> >> freeswitch at voipserverServerFreeswitch> > >>>> >> freeswitch at voipserverServerFreeswitch> 2009-04-30 17:52:49 > [NOTICE] > >>>> >> switch_channel.c:602 switch_channel_set_name() New Channel > >>>> sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [fc670e69-1143-4241-8364-3158f1ffa6ef] > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2912 > sofia_handle_sip_i_state() > >>>> >> Channel > >>>> >> sofia/external/07771236762 at sipgate.co.uk entering state > [received][100] > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:2919 > sofia_handle_sip_i_state() > >>>> Remote > >>>> >> SDP: > >>>> >> v=0 > >>>> >> o=root 15141 15141 IN IP4 217.10.66.71 > >>>> >> s=session > >>>> >> c=IN IP4 217.10.66.71 > >>>> >> t=0 0 > >>>> >> m=audio 12950 RTP/AVP 8 0 3 97 18 112 101 > >>>> >> a=rtpmap:8 PCMA/8000 > >>>> >> a=rtpmap:0 PCMU/8000 > >>>> >> a=rtpmap:3 GSM/8000 > >>>> >> a=rtpmap:97 iLBC/8000 > >>>> >> a=fmtp:97 mode=30 > >>>> >> a=rtpmap:18 G729/8000 > >>>> >> a=fmtp:18 annexb=no > >>>> >> a=rtpmap:112 G726-32/8000 > >>>> >> a=rtpmap:101 telephone-event/8000 > >>>> >> a=fmtp:101 0-16 > >>>> >> a=silenceSupp:off - - - - > >>>> >> a=ptime:20 > >>>> >> > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > >>>> sofia_glue_negotiate_sdp() > >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:98:8000:20] > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > >>>> sofia_glue_negotiate_sdp() > >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[SPEEX:99:16000:20] > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > >>>> sofia_glue_negotiate_sdp() > >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2931 > >>>> sofia_glue_negotiate_sdp() > >>>> >> Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:1912 > >>>> sofia_glue_tech_set_codec() > >>>> >> Set Codec sofia/external/07771236762 at sipgate.co.uk PCMA/8000 20 ms > 160 > >>>> >> samples > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia_glue.c:2891 > >>>> sofia_glue_negotiate_sdp() > >>>> >> Set 2833 dtmf payload to 101 > >>>> >> 2009-04-30 17:52:49 [DEBUG] sofia.c:3078 > sofia_handle_sip_i_state() > >>>> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_NEW -> > >>>> CS_INIT > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [BREAK] > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> >> Running State Change CS_INIT > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> INIT > >>>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:83 sofia_on_init() > >>>> sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk SOFIA INIT > >>>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:111 sofia_on_init() > >>>> >> (sofia/external/07771236762 at sipgate.co.uk) State Change CS_INIT -> > >>>> >> CS_ROUTING > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [BREAK] > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:480 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> INIT going to sleep > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> >> Running State Change CS_ROUTING > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:483 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> ROUTING > >>>> >> 2009-04-30 17:52:49 [DEBUG] mod_sofia.c:130 sofia_on_routing() > >>>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA ROUTING > >>>> >> 2009-04-30 17:52:49 [DEBUG] switch_core_state_machine.c:78 > >>>> >> switch_core_standard_on_routing() > >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard ROUTING > >>>> >> 2009-04-30 17:52:49 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > >>>> >> Processing 07771236762->00442083324655 in context public > >>>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk parsing > >>>> >> [public->skype_uri] continue=false > >>>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Regex (PASS) > >>>> >> [skype_uri] destination_number(00442083324655) =~ > /^(00442083324655)$/ > >>>> >> break=on-false > >>>> >> Dialplan: sofia/external/07771236762 at sipgate.co.uk Action > >>>> >> bridge(skypiax/skypiax1/xyzTestUK) > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:114 > >>>> >> switch_core_standard_on_routing() (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> ROUTING going to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> >> Running State Change CS_EXECUTE > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> EXECUTE > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() > >>>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA EXECUTE > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:151 > >>>> >> switch_core_standard_on_execute() > >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard EXECUTE > >>>> >> EXECUTE > >>>> >> > >>>> > sofia/external/07771236762 at sipgate.co.ukbridge(skypiax/skypiax1/xyzTestUK) > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:585 > >>>> channel_outgoing_channel() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?585 ?][ ? ? ? ? > ?][-1, 0, 0] > >>>> >> globals.SKYPIAX_INTERFACES[1].name=|||skypiax1|||? > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:151 skypiax_tech_init() > rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?151 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> skypiax_codec > >>>> >> SUCCESS > >>>> >> 2009-04-30 17:52:51 [NOTICE] switch_channel.c:602 > >>>> >> switch_channel_set_name() > >>>> >> New Channel skypiax/skypiax1/xyzTestUK > >>>> >> [0375c668-b4a2-4364-a8c6-0a718d4f00a3] > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:773 skypiax_call() > rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?773 ?][skypiax1 ?][-1, 0, > 0] Calling > >>>> >> Skype, rdest is: xyzTestUK > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE > ?1262 > >>>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AGC OFF|||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE > ?1262 > >>>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||SET AEC OFF|||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE > ?1262 > >>>> >> ][skypiax1 ?][-1, 0, 0] SENDING: |||CALL xyzTestUK|||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:642 > >>>> channel_outgoing_channel() > >>>> >> (skypiax/skypiax1/xyzTestUK) State Change CS_NEW -> CS_INIT > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 > channel_kill_channel() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running > State > >>>> >> Change > >>>> >> CS_INIT > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:177 channel_on_init() > >>>> >> (skypiax/skypiax1/xyzTestUK) State Change CS_INIT -> CS_ROUTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 > channel_kill_channel() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:182 channel_on_init() > rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?182 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL INIT > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:480 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State INIT > going > >>>> to > >>>> >> sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running > State > >>>> >> Change > >>>> >> CS_ROUTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > ROUTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:257 channel_on_routing() > rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?257 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL ROUTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:63 > >>>> >> originate_on_routing() (skypiax/skypiax1/xyzTestUK) State Change > >>>> >> CS_ROUTING > >>>> >> -> CS_CONSUME_MEDIA > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 > channel_kill_channel() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, 0, > 0] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:483 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > ROUTING > >>>> going > >>>> >> to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running > State > >>>> >> Change > >>>> >> CS_CONSUME_MEDIA > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > >>>> CONSUME_MEDIA > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||AGC OFF||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||AEC OFF||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 0, 0] > >>>> >> READING: > >>>> >> |||CALL 455 STATUS UNPLACED||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, > 0, 0] > >>>> Skype > >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: > >>>> >> UNPLACED,where: > >>>> >> NULL! > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:371 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?371 ?][skypiax1 ?][-1, > 3,116] > >>>> >> skype_call: 455 is now UNPLACED > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM > cards.pcm.hdmi > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 3,116] > >>>> >> READING: > >>>> >> |||CALL 455 STATUS ROUTING||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, > 3,116] > >>>> Skype > >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: > >>>> >> ROUTING,where: > >>>> >> NULL! > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:365 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?365 ?][skypiax1 ?][-1, > 3,117] > >>>> >> skype_call: 455 is now ROUTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 3,117] > >>>> >> READING: > >>>> >> |||CALL 455 FAILUREREASON 7||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, > 3,117] > >>>> Skype > >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: FAILUREREASON, > value: > >>>> >> 7,where: > >>>> >> NULL! > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:201 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?201 ?][skypiax1 ?][-1, > 3,117] > >>>> Skype > >>>> >> FAILED on skype_call 455. Let's wait for the FAILED message. > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 3,117] > >>>> >> READING: > >>>> >> |||CALL 455 VAA_INPUT_STATUS FALSE||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, > 3,117] > >>>> Skype > >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: VAA_INPUT_STATUS, > value: > >>>> >> FALSE,where: NULL! > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 3,117] > >>>> >> READING: > >>>> >> |||CALL 455 STATUS FAILED||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:167 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?167 ?][skypiax1 ?][-1, > 3,117] > >>>> Skype > >>>> >> MSG: message: CALL, obj: CALL, id: 455, prop: STATUS, value: > >>>> FAILED,where: > >>>> >> NULL! > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:334 > >>>> >> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?334 ?][skypiax1 ?][-1, > 3,112] we > >>>> >> tried > >>>> >> to call Skype on skype_call 455 and Skype has now FAILED > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> 672 > >>>> >> ?][skypiax1 ?][-1, 1,112] skype call ended > >>>> >> 2009-04-30 17:52:51 [NOTICE] mod_skypiax.c:680 > >>>> >> skypiax_signaling_thread_func() Hangup skypiax/skypiax1/xyzTestUK > >>>> >> [CS_CONSUME_MEDIA] [NORMAL_CLEARING] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > >>>> >> switch_channel_perform_hangup() Send signal > skypiax/skypiax1/xyzTestUK > >>>> >> [KILL] > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:293 > channel_kill_channel() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?293 ?][skypiax1 ?][-1, > 1,112] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_KILL > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 > channel_kill_channel() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, > 1,112] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_ivr_originate.c:2086 > >>>> >> switch_ivr_originate() Originate Resulted in Error Cause: 16 > >>>> >> [NORMAL_CLEARING] > >>>> >> 2009-04-30 17:52:51 [INFO] mod_dptools.c:2074 > audio_bridge_function() > >>>> >> Originate Failed. ?Cause: NORMAL_CLEARING > >>>> >> 2009-04-30 17:52:51 [NOTICE] mod_dptools.c:2106 > audio_bridge_function() > >>>> >> Hangup sofia/external/07771236762 at sipgate.co.uk [CS_EXECUTE] > >>>> >> [NORMAL_CLEARING] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_channel.c:1641 > >>>> >> switch_channel_perform_hangup() Send signal sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [KILL] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:490 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> EXECUTE going to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> >> Running State Change CS_HANGUP > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> HANGUP > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:323 sofia_on_hangup() > Channel > >>>> >> sofia/external/07771236762 at sipgate.co.uk hanging up, cause: > >>>> >> NORMAL_CLEARING > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_sofia.c:399 sofia_on_hangup() > >>>> Responding > >>>> >> to > >>>> >> INVITE with: 480 > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > >>>> >> switch_core_standard_on_hangup() > >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard HANGUP, cause: > >>>> >> NORMAL_CLEARING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> HANGUP going to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> Change CS_HANGUP -> CS_REPORTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> >> Running State Change CS_REPORTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > >>>> >> switch_core_session_reporting_state() (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) State REPORTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > >>>> CONSUME_MEDIA > >>>> >> going to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running > State > >>>> >> Change > >>>> >> CS_HANGUP > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > HANGUP > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() > rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?228 ?][skypiax1 ?][-1, > 1,112] hanging > >>>> up > >>>> >> skype call: 455 > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262 > >>>> >> skypiax_signaling_write() rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE > ?1262 > >>>> >> ][skypiax1 ?][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP|||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() > rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?235 ?][skypiax1 ?][-1, > 1,112] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46 > >>>> >> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK > Standard > >>>> >> HANGUP, > >>>> >> cause: NORMAL_CLEARING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > HANGUP > >>>> going > >>>> >> to > >>>> >> sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > Change > >>>> >> CS_HANGUP -> CS_REPORTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927 > >>>> >> switch_core_session_signal_state_change() Send signal > >>>> >> skypiax/skypiax1/xyzTestUK [BREAK] > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 > channel_kill_channel() > >>>> rev > >>>> >> 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?300 ?][skypiax1 ?][-1, > 1,112] > >>>> >> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running > State > >>>> >> Change > >>>> >> CS_REPORTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > >>>> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) > >>>> State > >>>> >> REPORTING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53 > >>>> >> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK > Standard > >>>> >> REPORTING, cause: NORMAL_CLEARING > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609 > >>>> >> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) > >>>> State > >>>> >> REPORTING going to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410 > >>>> >> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State > Change > >>>> >> CS_REPORTING -> CS_DESTROY > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061 > >>>> >> switch_core_session_thread() Session 2 > (skypiax/skypiax1/xyzTestUK) > >>>> >> Locked, > >>>> >> Waiting on external entities > >>>> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079 > >>>> >> switch_core_session_thread() Session 2 > (skypiax/skypiax1/xyzTestUK) > >>>> Ended > >>>> >> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081 > >>>> >> switch_core_session_thread() Close Channel > skypiax/skypiax1/xyzTestUK > >>>> >> [CS_DESTROY] > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > >>>> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) > State > >>>> >> DESTROY > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60 > >>>> >> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK > Standard > >>>> >> DESTROY > >>>> >> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559 > >>>> >> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) > State > >>>> >> DESTROY going to sleep > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?57 ? ][skypiax1 ?][-1, > 1,112] > >>>> >> READING: > >>>> >> |||ERROR 559 CALL: Action failed||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?91 ? ][skypiax1 ?][-1, > 1,112] > >>>> Skype > >>>> >> got ERROR: |||ERROR||| > >>>> >> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 > >>>> skypiax_signaling_read() > >>>> >> rev 13177[(nil)|37 ? ? ][DEBUG_SKYPE ?93 ? ][skypiax1 ?][-1, > 1,110] > >>>> >> skype_call now is DOWN > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672 > >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> 672 > >>>> >> ?][skypiax1 ?][-1, 1,110] skype call ended > >>>> >> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687 > >>>> >> skypiax_signaling_thread_func() rev 13177[(nil)|37 ? ? > ][DEBUG_SKYPE > >>>> 687 > >>>> >> ?][skypiax1 ?][-1, 1,110] no session > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53 > >>>> >> switch_core_standard_on_reporting() sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk Standard REPORTING, cause: > NORMAL_CLEARING > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609 > >>>> >> switch_core_session_reporting_state() (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) State REPORTING going to sleep > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410 > >>>> >> switch_core_session_run() > (sofia/external/07771236762 at sipgate.co.uk) > >>>> State > >>>> >> Change CS_REPORTING -> CS_DESTROY > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061 > >>>> >> switch_core_session_thread() Session 1 (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) Locked, Waiting on external entities > >>>> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079 > >>>> >> switch_core_session_thread() Session 1 (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) Ended > >>>> >> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081 > >>>> >> switch_core_session_thread() Close Channel sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk [CS_DESTROY] > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > >>>> >> switch_core_session_destroy_state() (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) State DESTROY > >>>> >> 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > >>>> >> sofia/external/07771236762 at sipgate.co.uk SOFIA DESTROY > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60 > >>>> >> switch_core_standard_on_destroy() > >>>> >> sofia/external/07771236762 at sipgate.co.ukStandard DESTROY > >>>> >> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559 > >>>> >> switch_core_session_destroy_state() (sofia/external/ > >>>> >> 07771236762 at sipgate.co.uk) State DESTROY going to sleep > >>>> >> -- > >>>> >> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > >>>> >> Telefonanschluss f?r nur 17,95 Euro/mtl.!* > >>>> >> > http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a > >>>> >> > >>>> >> _______________________________________________ > >>>> >> Freeswitch-users mailing list > >>>> >> Freeswitch-users at 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 at hotmail.com > > >>>> > > >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >>>> > IRC: irc.freenode.net #freeswitch > >>>> > > >>>> > FreeSWITCH Developer Conference > >>>> > sip:888 at conference.freeswitch.org > > >>>> > iax:guest at conference.freeswitch.org/888 > >>>> > > >>>> > googletalk:conf+888 at conference.freeswitch.org > >>>> > pstn:213-799-1400 > >>>> > > >>>> > >>>> -- > >>>> Sent from my mobile device > >>>> > >>>> Sincerely, > >>>> > >>>> Giovanni Maruzzelli > >>>> ========================================= > >>>> www.celliax.org > >>>> via Pierlombardo 9, 20135 Milano > >>>> Italy > >>>> gmaruzz at celliax dot org > >>>> Cell : +39-347-2665618 > >>>> Fax : +39-02-87390039 > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>> > >>> -- > >>> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01 From mszlazak at aol.com Fri May 1 16:03:24 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 01 May 2009 19:03:24 -0400 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com> Message-ID: <8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com> Hi Moiz, I've checking out mod_pocketshinx against other asr's on Windows with the same hardware. No matter what settings one tries, mod_pocketsphinx is virtually unusable in real world scenarios. One can play around with mod_pocketsphinx settings so that it picks voice up well but then there better not be any background noise either from a bad connection or just everyday sounds. It just way to sensitive and of couse you'll notice this problem most with cell phones. If you adjust the settings to try blocking out background noise you'll find you don't suceed all that well and then there are problems picking up the callers voice. It looks like some kind of signal pre-processing is required that isn't in place yet but we all know that this is a work-in progress. I don't know if esl would make any difference. To use FS and an ASR/TTS I just bridge calls to another ASR application for now. Mark -----Original Message----- From: Moiz Chinoy To: freeswitch-users at lists.freeswitch.org Sent: Fri, 1 May 2009 6:38 am Subject: [Freeswitch-users] Ways of Integrating Sphinx... Hi, I know only two ways of Sphinx - FS integration and its through mod_pocketsphinx and ESL. Performance with mod_pocketsphinx was not very good especially prompts were not playing properly. I haven't tried ESL. Can anyone guide what are other possibilities and which one is best in stability and can any one be deployed in live environment. -- Regards, Moiz Chinoy. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/d9243c20/attachment.html From q.edward at gmail.com Fri May 1 17:26:53 2009 From: q.edward at gmail.com (Edward Q.) Date: Fri, 1 May 2009 20:26:53 -0400 Subject: [Freeswitch-users] Asterisk AGI.PL Total Noob HELP plz... Message-ID: <89313a90905011726o74d525abve6d74c641f741721@mail.gmail.com> Hi guys .. I am a total noob on all this FS and asterisk thing .. I have a system already running on Asterisk and i want to migrate it to FS I have an AGI made in perl that runs everytime and i need to be at least pointed out on how i can approach on transferring this to FS Here is the AGI .. its simple ... #!/usr/bin/perl use Asterisk::AGI; use DBI; use strict; my $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $input{'callerid'} =~ /(^.+<(\d+)>$)|((^\d+$))/; $input{'calleridani'} = $2 || $3; #my $userid = $input{'calleridani'}; my $userid=$input{'extension'}; #my $userid='1001'; # Config options my %MYSQL = ( hostname => "localhost", username => "callmeuser", password => "mycallmepass", database => "executives" ); my $dbh = DBI->connect("dbi:mysql:$MYSQL{database}:$MYSQL{hostname}","$MYSQL{username}","$MYSQL{password}")|| die("Couldn't connect to database!\n"); #============ $AGI->verbose("Connected to database."); $AGI->verbose("Call for : $userid"); #====== print STDERR "$userid"; print STDERR "hello testing db connected"; my $debug=2; my $phone=""; #debug("Connect to database"); #debug("Transferred call, using original cid: $name",5); my $str ="select phone from timetable where exten='$userid' and online='Y' "; $AGI->verbose("Checking if $userid is online right now and get his phone number from db."); $AGI->verbose("$str"); print STDERR "$str"; my $sth =$dbh->prepare($str); $sth->execute || die("Couldn't exec sth2!"); #my $pin = $sth->fetchrow_hashref; #print STDERR "xxxx"; #print STDERR "$pin"; while (my @row = $sth->fetchrow_array) { $phone = $row[0]; print STDERR "xxxx"; print STDERR "$phone"; print STDERR "vvvvx"; $AGI->verbose("$userid is online and his phone number is $phone"); #$AGI->exec('DIAL', "SIP/$phone"); } if ($phone!="") { $AGI->verbose("Calling $userid at $phone"); my $dialstr="SIP/$phone"."@"."209.9.9.34"; $AGI->set_callerid(7347777777); $AGI->exec('DIAL', $dialstr); my $st=$AGI->get_full_variable('status',$dialstr); $AGI->verbose("channel status is : $st"); } else { $AGI->verbose("$userid is not online."); #$AGI->stream_file('custom/myrecording'); $AGI->stream_file('followme/sorry'); $AGI->stream_file('en/vm-nobodyavail'); } $AGI->hangup(); Thanks everyone for all the help. Ed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090501/ea1dc439/attachment.html From diego.viola at gmail.com Sat May 2 06:20:45 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 2 May 2009 09:20:45 -0400 Subject: [Freeswitch-users] Asterisk AGI.PL Total Noob HELP plz... In-Reply-To: <89313a90905011726o74d525abve6d74c641f741721@mail.gmail.com> References: <89313a90905011726o74d525abve6d74c641f741721@mail.gmail.com> Message-ID: <86a32abc0905020620x62396be7j79a18523968f98e8@mail.gmail.com> Hi Edward, You could use perl from the dialplan or use the Event socket, or ESL, I use the event socket with a ruby library called freeswitcher and it works really well. Just take a look at the wiki (http://wiki.freeswitch.org/) for more info and/or ask us for help at #freeswitch @irc.freenode.net. Hope that gives some idea to you. Regards, Diego On Fri, May 1, 2009 at 8:26 PM, Edward Q. wrote: > Hi guys .. > > I am a total noob on all this FS and asterisk thing .. > I have a system already running on Asterisk and i want to migrate it to FS > I have an AGI made in perl that runs everytime and i need to be at least > pointed out on how i can approach on transferring this to FS > > Here is the AGI .. its simple ... > > > #!/usr/bin/perl > use Asterisk::AGI; > use DBI; > use strict; > > my $AGI = new Asterisk::AGI; > my %input = $AGI->ReadParse(); > $input{'callerid'} =~ /(^.+<(\d+)>$)|((^\d+$))/; > $input{'calleridani'} = $2 || $3; > #my $userid = $input{'calleridani'}; > my $userid=$input{'extension'}; > #my $userid='1001'; > # Config options > my %MYSQL = ( > ??? hostname??? =>??? "localhost", > ??? username??? =>??? "callmeuser", > ??? password??? =>??? "mycallmepass", > ??? database??? =>??? "executives" > ); > > my $dbh = > DBI->connect("dbi:mysql:$MYSQL{database}:$MYSQL{hostname}","$MYSQL{username}","$MYSQL{password}")|| > die("Couldn't connect to database!\n"); > #============ > $AGI->verbose("Connected to database."); > $AGI->verbose("Call for : $userid"); > #====== > > > ?print STDERR "$userid"; > ?print STDERR "hello testing db connected"; > my $debug=2; > my $phone=""; > #debug("Connect to database"); > #debug("Transferred call, using original cid: $name",5); > my $str ="select phone from timetable where exten='$userid' and online='Y' > "; > $AGI->verbose("Checking if $userid is online right now and get his phone > number from db."); > $AGI->verbose("$str"); > print STDERR "$str"; > my $sth =$dbh->prepare($str); > $sth->execute || die("Couldn't exec sth2!"); > #my $pin = $sth->fetchrow_hashref; > #print STDERR "xxxx"; > #print STDERR "$pin"; > > while (my @row = $sth->fetchrow_array) > { > $phone = $row[0]; > print STDERR "xxxx"; > print STDERR "$phone"; > print STDERR "vvvvx"; > $AGI->verbose("$userid is online and his phone number is $phone"); > #$AGI->exec('DIAL', "SIP/$phone"); > ?} > if? ($phone!="") > > { > $AGI->verbose("Calling? $userid at $phone"); > my $dialstr="SIP/$phone"."@"."209.9.9.34"; > $AGI->set_callerid(7347777777); > > $AGI->exec('DIAL', $dialstr); > my $st=$AGI->get_full_variable('status',$dialstr); > $AGI->verbose("channel status is : $st"); > } > > else > > { > $AGI->verbose("$userid is not online."); > #$AGI->stream_file('custom/myrecording'); > $AGI->stream_file('followme/sorry'); > $AGI->stream_file('en/vm-nobodyavail'); > } > > $AGI->hangup(); > > > Thanks everyone for all the help. > Ed > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sat May 2 07:42:38 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 2 May 2009 09:42:38 -0500 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com> <8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com> Message-ID: <069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org> On May 1, 2009, at 6:03 PM, mszlazak at aol.com wrote: > Hi Moiz, > > I've checking out mod_pocketshinx against other asr's on Windows > with the same hardware. > No matter what settings one tries, mod_pocketsphinx is virtually > unusable in real world scenarios. I have used it and it works fine... I think your expectations are a bit high for it... Complex things like dictation is not what PocketSphinx is for. You should try linux cuz I know it works great there. > One can play around with mod_pocketsphinx settings so that it picks > voice up well but then there better not be any background noise > either from a bad connection or just everyday sounds. There is no other ASR out there that doesn't get pissed off at background noise or any noise for that matter... have you called AT&T and Sprint lately? My dogs barking in the background really send theirs into fits and they paid tons of money for it. > It just way to sensitive and of couse you'll notice this problem > most with cell phones. Same with commercial ASR, Granted the acoustical model for PocketSphinx wasn't done with any files recorded from cellphone from what I can tell. You can do adaptation of the acoustical model as per the Sphinx wiki to make it more accurate for your needs.... that takes time and effort but it works. > If you adjust the settings to try blocking out background noise > you'll find you don't suceed all that well and then there are > problems picking up the callers voice. Those settings are for telling when the person stopped talking... nothing more. > It looks like some kind of signal pre-processing is required that > isn't in place yet but we all know that this is a work-in progress. I'm not working on it... I run the pizza demo with PS and it works from my polycom rather well I would say it gets some things wrong but it does score them low so you can verify it in your scripts. > I don't know if esl would make any difference. To use FS and an ASR/ > TTS I just bridge calls to another ASR application for now. > > Mark Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090502/6ca0280d/attachment-0001.html From mszlazak at aol.com Sat May 2 10:13:00 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 02 May 2009 13:13:00 -0400 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com> <069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org> Message-ID: <8CB99606108C2AA-964-2933@WEBMAIL-DY37.sysops.aol.com> In my comments on mod_pocketsphinx, I wasn't clear enough about it being "virtually unusable in real world scenarios." Also, the grammars I'm talking about are either single words, like "yes/no" or more complex like "leave a message." It doesn't matter how complex the grammar, the issue remains. My comments are meant in comparison to other asr's and in everyday situations of background noise. I'm not taliking about checking things out at a concert, race track, subway, construction project, etc. When compared to my AT&T 411 service, AT&T's asr has no where near the problems in dealing with the background noises I'm talking about and is very usable in the real world situations I'm taliking about. Moreover, when comparing to another vendors asr on my hardware then that vendors asr also has no were near the problems mod_pocketsphinx has and again is very usable in a real world situation. That's why I suggested using something other than mod_pocketsphinx. I think that mod_pocketspinx is not able to deal with low signal-to-noise ratios to the point where it can be used in telephony at all. At least that's the way it seems to me. I don't know what else to say. That's been my experience with mod_pocketsphinx ?? Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Sat, 2 May 2009 7:42 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... On May 1, 2009, at 6:03 PM, mszlazak at aol.com wrote: Hi Moiz, I've checking out mod_pocketshinx against other asr's on Windows with the same hardware.? No matter what settings one tries, mod_pocketsphinx is virtually unusable in real world scenarios.? I have used it and it works fine... I think your expectations are a bit high for it... Complex things like dictation is not what PocketSphinx is for. ?You should try linux cuz I know it works great there. One can play around with mod_pocketsphinx settings so that it picks voice up well but then there better not be any background noise either from a bad connection or just everyday sounds.? There is no other ASR out there that doesn't get pissed off at background noise or any noise for that matter... have you called AT&T and Sprint lately? ?My dogs barking in the background really send theirs into fits and they paid tons of money for it. ? It just way to sensitive and of couse you'll notice this problem most with cell phones. Same with commercial ASR, Granted the acoustical model for PocketSphinx wasn't done with any files recorded from cellphone from what I can tell. ?You can do adaptation of the acoustical model as per the Sphinx wiki to make it more accurate for your needs.... that takes time and effort but it works. If you adjust the settings to try blocking out background noise you'll find you don't suceed all that well and then there are problems picking up the callers voice. Those settings are for telling when the person stopped talking... nothing more. It looks like some kind of signal pre-processing is required that isn't in place yet but we all know that this is a work-in progress. I'm not working on it... I run the pizza demo with PS and it works from my polycom rather well I would say it gets some things wrong but it does score them low so you can verify it in your scripts. I don't know if esl would make any difference. To use FS and an ASR/TTS I just bridge calls to another ASR application for now.? Mark Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090502/f4d6aa80/attachment.html From austad at signal15.com Sat May 2 11:45:36 2009 From: austad at signal15.com (Jay Austad) Date: Sat, 2 May 2009 13:45:36 -0500 Subject: [Freeswitch-users] t.38 fax error Message-ID: <6244F582-3FD6-4118-8857-6354545C3CFA@signal15.com> I compiled mod_fax and enabled it in the freeswitch config. I tried sending a test fax to 9978 and 9979, and get this error for both: 2009-05-02 13:35:08 [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel sofia/internal/1001 at 10.128.0.10 [317b59c6-157b-479b-8fb0-f80453f67355] 2009-05-02 13:35:08 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Fax Extension->9978 in context default 2009-05-02 13:35:08 [NOTICE] mod_dptools.c:649 answer_function() Channel [sofia/internal/1001 at 10.128.0.10] has been answered 2009-05-02 13:35:09 [ERR] sofia.c:3217 sofia_handle_sip_i_state() Reinvite Codec Error! I'm using Zoiper to send the fax. Any ideas why this is failing? -- jay austad | 612.423.1433 | austad at signal15.com From nik.middleton at noblesolutions.co.uk Sat May 2 13:36:30 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 2 May 2009 21:36:30 +0100 Subject: [Freeswitch-users] Hang-up event - Alternative? Message-ID: Hi Guys, Is there an alternative to the hang-up event that doesn't send quite as much data? This event is HUGE! All I'm looking for this the result of the call, duration, dialed number and the ability to pass variables. The hang-up event does all of this I know, but I also get everything including the stock market reports (just kidding) Right now I'm using custom events for successful calls and the BACKGROUND_JOB for call fails as my application is running an lua script on call answer, but this doesn't get called if the call fails Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090502/2aa1ba77/attachment.html From mattdfong at gmail.com Sat May 2 14:21:49 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Sat, 2 May 2009 14:21:49 -0700 Subject: [Freeswitch-users] Hang-up event - Alternative? In-Reply-To: References: Message-ID: <4256bf830905021421l11aa1a2fhe45f3efe4ad2b533@mail.gmail.com> You can always have your lua script fire a custom event on api_hangup...this will only send the data you specify in your event. On Sat, May 2, 2009 at 1:36 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Is there an alternative to the hang-up event that doesn?t send quite as > much data? This event is HUGE! > > > > All I?m looking for this the result of the call, duration, dialed number > and the ability to pass variables. The hang-up event does all of this I > know, but I also get everything including the stock market reports (just > kidding) > > > > Right now I?m using custom events for successful calls and the > BACKGROUND_JOB for call fails as my application is running an lua script on > call answer, but this doesn?t get called if the call fails > > > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090502/d61674da/attachment-0001.html From nik.middleton at noblesolutions.co.uk Sat May 2 15:25:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 2 May 2009 23:25:51 +0100 Subject: [Freeswitch-users] Hang-up event - Alternative? In-Reply-To: <4256bf830905021421l11aa1a2fhe45f3efe4ad2b533@mail.gmail.com> References: <4256bf830905021421l11aa1a2fhe45f3efe4ad2b533@mail.gmail.com> Message-ID: That won't work unless I'm mistaken. Well it will if the call is answered, but if it fails, the lua script will not be called. So if the result is BUSY, the script won't be called. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthew Fong Sent: 02 May 2009 22:22 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hang-up event - Alternative? You can always have your lua script fire a custom event on api_hangup...this will only send the data you specify in your event. On Sat, May 2, 2009 at 1:36 PM, Nik Middleton wrote: Hi Guys, Is there an alternative to the hang-up event that doesn't send quite as much data? This event is HUGE! All I'm looking for this the result of the call, duration, dialed number and the ability to pass variables. The hang-up event does all of this I know, but I also get everything including the stock market reports (just kidding) Right now I'm using custom events for successful calls and the BACKGROUND_JOB for call fails as my application is running an lua script on call answer, but this doesn't get called if the call fails Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090502/38af9513/attachment.html From mrene_lists at avgs.ca Sat May 2 15:36:50 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 2 May 2009 18:36:50 -0400 Subject: [Freeswitch-users] Hang-up event - Alternative? In-Reply-To: References: Message-ID: <24559B4C-D088-4F78-B750-1BA5EE96F852@avgs.ca> You are looking into optimizing the wrong things. DId you experience any problems directly related to the hangup event containing extra data? Math On 2-May-09, at 4:36 PM, Nik Middleton wrote: > Hi Guys, > > Is there an alternative to the hang-up event that doesn?t send quite > as much data? This event is HUGE! > > All I?m looking for this the result of the call, duration, dialed > number and the ability to pass variables. The hang-up event does > all of this I know, but I also get everything including the stock > market reports (just kidding) > > Right now I?m using custom events for successful calls and the > BACKGROUND_JOB for call fails as my application is running an lua > script on call answer, but this doesn?t get called if the call fails > > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090502/00f928cb/attachment.html From diego.viola at gmail.com Sat May 2 16:57:22 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 2 May 2009 19:57:22 -0400 Subject: [Freeswitch-users] Ruby and ESL help Message-ID: <86a32abc0905021657s72ae5d42wb98fa1e1ba06b0f3@mail.gmail.com> Hello everyone, I was trying to test ESL with Ruby, and I made this: " require 'socket' require 'ESL' TCPServer.new('127.0.0.1', '8084') con = ESL::ESLconnection.new('127.0.0.1', '8084', '') con.execute('answer') con.execute('playback', '/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav') " I can connect from freeswitch with sync and async mode, but it doesn't do anything more than that, it doesn't execute my answer or playback, anyone knows what's wrong with it? I use the freeswitcher lib and it works great, but I also want to try ESL. Thanks, Diego From diego.viola at gmail.com Sat May 2 17:18:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 2 May 2009 20:18:20 -0400 Subject: [Freeswitch-users] Ruby and ESL help In-Reply-To: <86a32abc0905021657s72ae5d42wb98fa1e1ba06b0f3@mail.gmail.com> References: <86a32abc0905021657s72ae5d42wb98fa1e1ba06b0f3@mail.gmail.com> Message-ID: <86a32abc0905021718k61985a91j210a148890c493e6@mail.gmail.com> I'm trying to do Event socket outbound btw. On Sat, May 2, 2009 at 7:57 PM, Diego Viola wrote: > Hello everyone, > > I was trying to test ESL with Ruby, and I made this: > > " > require 'socket' > require 'ESL' > > TCPServer.new('127.0.0.1', '8084') > con = ESL::ESLconnection.new('127.0.0.1', '8084', '') > con.execute('answer') > con.execute('playback', > '/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav') > " > > I can connect from freeswitch with sync and async mode, but it doesn't > do anything more than that, it doesn't execute my answer or playback, > anyone knows what's wrong with it? I use the freeswitcher lib and it > works great, but I also want to try ESL. > > Thanks, > > Diego > From gk at exram.de Sun May 3 09:58:37 2009 From: gk at exram.de (Guido Kuth) Date: Sun, 3 May 2009 16:58:37 +0000 Subject: [Freeswitch-users] Re-2: Ruby and ESL help Message-ID: Hello Diego, I don't know ruby but I was playing around with outbound socket as well. You have to start your TCPServer and then listen for connections on port 8084 (if you want it like it is standard). If the TCPServer gets a connect request from FS you have to Accept the connection. In .NET this is TCPServer.Accept(). This Returns a TCPClient Object which represents a dedicated connection for this specific call. A new call creates a new TCPClient Object. After that you first have to send a Connect Message ("Connect\n\n") to FS. FS will answer immediately with all data belongig to the call. If this all ist done you can send an Answer command and/or whatever you want. Hope this helps...Guido Btw.: If you find out how one can handle real blocked execution of commands I would like to know how. I tried to playback a long file and my problem was that FS answers immediately after FS accepts the command to play this file, but there is nothing that will ever give you a notice about the playback has ended, what is an unsolved problem for me. -------- Original Message -------- Subject: Re: [Freeswitch-users] Ruby and ESL help (03-Mai-2009 2:23) From: Diego Viola To: gk at exram.de > I'm trying to do Event socket outbound btw. > > On Sat, May 2, 2009 at 7:57 PM, Diego Viola wrote: > > Hello everyone, > > > > I was trying to test ESL with Ruby, and I made this: > > > > " > > require 'socket' > > require 'ESL' > > > > TCPServer.new('127.0.0.1', '8084') > > con = ESL::ESLconnection.new('127.0.0.1', '8084', '') > > con.execute('answer') > > con.execute('playback', > > '/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav') > > " > > > > I can connect from freeswitch with sync and async mode, but it doesn't > > do anything more than that, it doesn't execute my answer or playback, > > anyone knows what's wrong with it? I use the freeswitcher lib and it > > works great, but I also want to try ESL. > > > > Thanks, > > > > Diego > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul.degt at gmail.com Sun May 3 10:51:00 2009 From: paul.degt at gmail.com (paul.degt) Date: Sun, 03 May 2009 13:51:00 -0400 Subject: [Freeswitch-users] Segfaults with core dump, how to handle Message-ID: <49FDD984.7060607@gmail.com> We experience sporadic seg faults in our production FS, version 1.0.3, load is very low, 10-15 users, runs under Centos 5.2 2.6.18-92.1.22.el5 SMP 64-bit. This is what I get in system log: May 3 10:39:01 hostname kernel: freeswitch[7578]: segfault at 0000000000000000 rip 00002aaab098e236 rsp 0000000040cdde50 error 4 Need advice on the best way to handle the situation, since it's our production switch I hesitate to use trunk version, we may hit some other unknown bugs. Which version is considered currently stable enough for production use? Thanks. From brian at freeswitch.org Sun May 3 10:56:04 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 3 May 2009 12:56:04 -0500 Subject: [Freeswitch-users] Segfaults with core dump, how to handle In-Reply-To: <49FDD984.7060607@gmail.com> References: <49FDD984.7060607@gmail.com> Message-ID: <3A390866-4404-4FF2-AA8A-9C83BD794774@freeswitch.org> Many bug fixes since 1.0.3 and SVN Trunk is what I would be using! On May 3, 2009, at 12:51 PM, paul.degt wrote: > We experience sporadic seg faults in our production FS, version 1.0.3, > load is very low, 10-15 users, runs under Centos 5.2 > 2.6.18-92.1.22.el5 > SMP 64-bit. This is what I get in system log: > May 3 10:39:01 hostname kernel: freeswitch[7578]: segfault at > 0000000000000000 rip 00002aaab098e236 rsp 0000000040cdde50 error 4 > > Need advice on the best way to handle the situation, since it's our > production switch I hesitate to use trunk version, we may hit some > other > unknown bugs. > Which version is considered currently stable enough for production > use? > > Thanks. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090503/bc98d4a9/attachment.html From kokoska.rokoska at post.cz Sun May 3 13:33:55 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Sun, 03 May 2009 22:33:55 +0200 Subject: [Freeswitch-users] FS & Outbound proxy Message-ID: <49FDFFB3.8050906@post.cz> Hi all, while I read some threads about Outbound Proxy, I'm still not sure how to use it :-) Well, what's going on: I want to send and receive calls from/to my TSP which uses outbound proxy. For that, I have to register with providers registrar (R1), receive calls from outbound proxy (O1) and send calls (D-URI) to outbound proxy (O1), but with R-URI pointing to real proxy (P1). BTW: registar and proxy challenge me for credentials... Registering with R1, receiving calls from O1 is simple and works fine. But I still cant successfuly send calls to O1 with R-URI of P1. When I try to add fs_path to "dialing" thru gateway, it is silently ignored... Any hint is really welcome :-) Best regards, kokoska.rokoska From brian at freeswitch.org Sun May 3 13:48:33 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 3 May 2009 15:48:33 -0500 Subject: [Freeswitch-users] FS & Outbound proxy In-Reply-To: <49FDFFB3.8050906@post.cz> References: <49FDFFB3.8050906@post.cz> Message-ID: <170EB8B9-4C33-4D97-82CA-D7635378D233@freeswitch.org> Setting the proxy vs register-proxy in the gateway should do what you want can you verify that? /b On May 3, 2009, at 3:33 PM, kokoska rokoska wrote: > > Hi all, > > while I read some threads about Outbound Proxy, I'm still not sure how > to use it :-) > > Well, what's going on: > I want to send and receive calls from/to my TSP which uses outbound > proxy. > For that, I have to register with providers registrar (R1), receive > calls from outbound proxy (O1) and send calls (D-URI) to outbound > proxy > (O1), but with R-URI pointing to real proxy (P1). > BTW: registar and proxy challenge me for credentials... > > Registering with R1, receiving calls from O1 is simple and works fine. > But I still cant successfuly send calls to O1 with R-URI of P1. > When I try to add fs_path to "dialing" thru gateway, it is silently > ignored... > > Any hint is really welcome :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090503/23215d15/attachment.html From diego.viola at gmail.com Sun May 3 14:00:50 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:00:50 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: References: Message-ID: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> Yep, it works Guido. require 'socket' server = TCPServer.new(8084) loop do con = server.accept con.puts "connect\n\n" con.puts "sendmsg\ncall-command: execute\nexecute-app-name: answer\n\n" con.puts "sendmsg\ncall-command: execute\nexecute-app-name: playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" end Thanks for the tip =D On Sun, May 3, 2009 at 12:58 PM, Guido Kuth wrote: > Hello Diego, > > I don't know ruby but I was playing around with outbound socket as well. You have to start your TCPServer and then listen for connections on port 8084 (if you want it like it is standard). If the TCPServer gets a connect request from FS you have to Accept the connection. In .NET this is TCPServer.Accept(). This Returns a TCPClient Object which represents a dedicated connection for this specific call. A new call creates a new TCPClient Object. After that you first have to send a Connect Message ("Connect\n\n") to FS. FS will answer immediately with all data belongig to the call. > > If this all ist done you can send an Answer command and/or whatever you want. > > Hope this helps...Guido > > Btw.: If you find out how one can handle real blocked execution of commands I would like to know how. I tried to playback a long file and my problem was that FS answers immediately after FS accepts the command to play this file, but there is nothing that will ever give you a notice about the playback has ended, what is an unsolved problem for me. > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Ruby and ESL help (03-Mai-2009 2:23) > From: ? ?Diego Viola > To: ? ? ?gk at exram.de > >> I'm trying to do Event socket outbound btw. >> >> On Sat, May 2, 2009 at 7:57 PM, Diego Viola wrote: >> > Hello everyone, >> > >> > I was trying to test ESL with Ruby, and I made this: >> > >> > " >> > require 'socket' >> > require 'ESL' >> > >> > TCPServer.new('127.0.0.1', '8084') >> > con = ESL::ESLconnection.new('127.0.0.1', '8084', '') >> > con.execute('answer') >> > con.execute('playback', >> > '/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav') >> > " >> > >> > I can connect from freeswitch with sync and async mode, but it doesn't >> > do anything more than that, it doesn't execute my answer or playback, >> > anyone knows what's wrong with it? I use the freeswitcher lib and it >> > works great, but I also want to try ESL. >> > >> > Thanks, >> > >> > Diego >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Sun May 3 14:06:36 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 3 May 2009 16:06:36 -0500 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> Message-ID: This is how we do it in perl with ESL... it should be very similar in Ruby. You shouldn't have to manually use sendmsg if you tie the fd from the socket to ESL like we do in perl. /b require ESL; use IO::Socket::INET; my $ip = "127.0.0.1"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8040', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); print $info->serialize(); my $uuid = $info->getHeader("unique-id"); $con->execute("answer", "", $uuid); $con->execute("playback", "/ram/swimp.raw", $uuid); close($new_sock); } On May 3, 2009, at 4:00 PM, Diego Viola wrote: > Yep, it works Guido. > > require 'socket' > > server = TCPServer.new(8084) > loop do > con = server.accept > con.puts "connect\n\n" > con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > answer\n\n" > con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" > end > > Thanks for the tip =D Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090503/ac0d2373/attachment.html From diego.viola at gmail.com Sun May 3 14:17:10 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:17:10 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> Message-ID: <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> I tried to use ESL::ESLconnection in ruby but I get this. [diego at localhost ruby]$ ruby test.rb test.rb:7:in `initialize': Wrong arguments for overloaded method 'ESLconnection.new'. (ArgumentError) Possible C/C++ prototypes are: ESLconnection.new(char const *host, char const *port, char const *password) ESLconnection.new(int socket) from test.rb:7:in `new' from test.rb:7 from test.rb:5:in `loop' from test.rb:5 [diego at localhost ruby]$ I made something like this: esl = ESL::ESLconnection.new(con) Where con is the accepted socket... should that work? Or do I have to specify host/port/password on the ESLconnection? Thanks, Diego On Sun, May 3, 2009 at 5:06 PM, Brian West wrote: > This is how we do it in perl with ESL... it should be very similar in Ruby. > You shouldn't have to manually use sendmsg if you tie the fd from the socket > to ESL like we do in perl. > /b > > require ESL; > use IO::Socket::INET; > my $ip = "127.0.0.1"; > my $sock = new IO::Socket::INET ( LocalHost => $ip, ?LocalPort => '8040', > ?Proto => 'tcp', ?Listen => 1, ?Reuse => 1 ); > die "Could not create socket: $!\n" unless $sock; > for(;;) { > ??my $new_sock = $sock->accept(); > ??my $pid = fork(); > ??if ($pid) { > ?? ?close($new_sock); > ?? ?next; > ??} > ??my $host = $new_sock->sockhost(); > ??my $fd = fileno($new_sock); > > ??my $con = new ESL::ESLconnection($fd); > ??my $info = $con->getInfo(); > ??print $info->serialize(); > ??my $uuid = $info->getHeader("unique-id"); > ??$con->execute("answer", "", $uuid); > ??$con->execute("playback", "/ram/swimp.raw", $uuid); > ??close($new_sock); > } > > > On May 3, 2009, at 4:00 PM, Diego Viola wrote: > > Yep, it works Guido. > > require 'socket' > > server = TCPServer.new(8084) > loop do > ???????con = server.accept > ???????con.puts "connect\n\n" > ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > answer\n\n" > ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" > end > > Thanks for the tip =D > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sun May 3 14:25:51 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 3 May 2009 16:25:51 -0500 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> Message-ID: <25108F11-A7C7-45AD-9EDB-89FFB12B2A18@freeswitch.org> You have to pass it the file descriptor I suspect like we do in perl, python and lua. /b On May 3, 2009, at 4:17 PM, Diego Viola wrote: > > esl = ESL::ESLconnection.new(con) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090503/9cd6c53d/attachment.html From diego.viola at gmail.com Sun May 3 14:26:13 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:26:13 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> Message-ID: <86a32abc0905031426n7e2dfbe6m7aeeaac0f7a4d770@mail.gmail.com> Do I need to do something with the file descriptor or fileno first? Sorry, I don't know perl. Diego On Sun, May 3, 2009 at 5:17 PM, Diego Viola wrote: > I tried to use ESL::ESLconnection in ruby but I get this. > > [diego at localhost ruby]$ ruby test.rb > test.rb:7:in `initialize': Wrong arguments for overloaded method > 'ESLconnection.new'. (ArgumentError) > Possible C/C++ prototypes are: > ? ?ESLconnection.new(char const *host, char const *port, char const *password) > ? ?ESLconnection.new(int socket) > ? ? ? ?from test.rb:7:in `new' > ? ? ? ?from test.rb:7 > ? ? ? ?from test.rb:5:in `loop' > ? ? ? ?from test.rb:5 > [diego at localhost ruby]$ > > I made something like this: > > esl = ESL::ESLconnection.new(con) > > Where con is the accepted socket... should that work? Or do I have to > specify host/port/password on the ESLconnection? > > Thanks, > > Diego > > On Sun, May 3, 2009 at 5:06 PM, Brian West wrote: >> This is how we do it in perl with ESL... it should be very similar in Ruby. >> You shouldn't have to manually use sendmsg if you tie the fd from the socket >> to ESL like we do in perl. >> /b >> >> require ESL; >> use IO::Socket::INET; >> my $ip = "127.0.0.1"; >> my $sock = new IO::Socket::INET ( LocalHost => $ip, ?LocalPort => '8040', >> ?Proto => 'tcp', ?Listen => 1, ?Reuse => 1 ); >> die "Could not create socket: $!\n" unless $sock; >> for(;;) { >> ??my $new_sock = $sock->accept(); >> ??my $pid = fork(); >> ??if ($pid) { >> ?? ?close($new_sock); >> ?? ?next; >> ??} >> ??my $host = $new_sock->sockhost(); >> ??my $fd = fileno($new_sock); >> >> ??my $con = new ESL::ESLconnection($fd); >> ??my $info = $con->getInfo(); >> ??print $info->serialize(); >> ??my $uuid = $info->getHeader("unique-id"); >> ??$con->execute("answer", "", $uuid); >> ??$con->execute("playback", "/ram/swimp.raw", $uuid); >> ??close($new_sock); >> } >> >> >> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >> >> Yep, it works Guido. >> >> require 'socket' >> >> server = TCPServer.new(8084) >> loop do >> ???????con = server.accept >> ???????con.puts "connect\n\n" >> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >> answer\n\n" >> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >> end >> >> Thanks for the tip =D >> >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! ?http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From diego.viola at gmail.com Sun May 3 14:26:45 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:26:45 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <25108F11-A7C7-45AD-9EDB-89FFB12B2A18@freeswitch.org> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> <25108F11-A7C7-45AD-9EDB-89FFB12B2A18@freeswitch.org> Message-ID: <86a32abc0905031426u3491decay2770d5a17156fe70@mail.gmail.com> Ok, I'll try that. Thanks. Diego On Sun, May 3, 2009 at 5:25 PM, Brian West wrote: > You have to pass it the file descriptor I suspect like we do in perl, python > and lua. > /b > On May 3, 2009, at 4:17 PM, Diego Viola wrote: > > esl = ESL::ESLconnection.new(con) > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kokoska.rokoska at post.cz Sun May 3 14:27:53 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Sun, 03 May 2009 23:27:53 +0200 Subject: [Freeswitch-users] FS & Outbound proxy In-Reply-To: <170EB8B9-4C33-4D97-82CA-D7635378D233@freeswitch.org> References: <49FDFFB3.8050906@post.cz> <170EB8B9-4C33-4D97-82CA-D7635378D233@freeswitch.org> Message-ID: <49FE0C59.3040102@post.cz> Brian West napsal(a): > Setting the proxy vs register-proxy in the gateway should do what you > want can you verify that? > Thank you very much, Brian, for you interest! Using proxy and regiter-proxy solves only 1 half of my problem => I can successfuly register with provider (register-proxy is in R-URI of REGISTER and packet is send to same address), but all calls (INVITEs) have in R-URI proxy address and are sent to this proxy instead of Outbnound proxy - becasue I have no idea how to tell to FreeSWITCH how to send INVITEs to OB. It is three independant machines - registrar, proxy, outbound proxy. What I found using sipsak is: 1. To be succesufly registered I can send REGISTER to registrar or to Outbound proxy, but R-URI of register should allways point to registrar. 2. For succesfull call I have to send INVITE to Otbound proxy with R-URI pointing to proxy. Best regards, kokoska.rokoska > /b > > On May 3, 2009, at 3:33 PM, kokoska rokoska wrote: > >> >> Hi all, >> >> while I read some threads about Outbound Proxy, I'm still not sure how >> to use it :-) >> >> Well, what's going on: >> I want to send and receive calls from/to my TSP which uses outbound proxy. >> For that, I have to register with providers registrar (R1), receive >> calls from outbound proxy (O1) and send calls (D-URI) to outbound proxy >> (O1), but with R-URI pointing to real proxy (P1). >> BTW: registar and proxy challenge me for credentials... >> >> Registering with R1, receiving calls from O1 is simple and works fine. >> But I still cant successfuly send calls to O1 with R-URI of P1. >> When I try to add fs_path to "dialing" thru gateway, it is silently >> ignored... >> >> Any hint is really welcome :-) >> >> Best regards, >> >> kokoska.rokoska >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sun May 3 14:27:55 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 3 May 2009 16:27:55 -0500 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031426n7e2dfbe6m7aeeaac0f7a4d770@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com> <86a32abc0905031426n7e2dfbe6m7aeeaac0f7a4d770@mail.gmail.com> Message-ID: What ever the equiv. function in ruby is. /b On May 3, 2009, at 4:26 PM, Diego Viola wrote: > Do I need to do something with the file descriptor or fileno first? > Sorry, I don't know perl. > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090503/e03e92cf/attachment.html From brian at freeswitch.org Sun May 3 14:29:22 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 3 May 2009 16:29:22 -0500 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> Message-ID: I think its con.fileno in this case? Not sure. /b On May 3, 2009, at 4:00 PM, Diego Viola wrote: > Yep, it works Guido. > > require 'socket' > > server = TCPServer.new(8084) > loop do > con = server.accept > con.puts "connect\n\n" > con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > answer\n\n" > con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" > end > > Thanks for the tip =D Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090503/fb6ca17c/attachment-0001.html From diego.viola at gmail.com Sun May 3 14:32:13 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:32:13 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> Message-ID: <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> NICE! It works, it works =D require 'socket' require 'ESL' server = TCPServer.new(8084) loop do con = server.accept fd = con.to_i esl = ESL::ESLconnection.new(fd) esl.execute('answer') esl.execute('playback', 'tone_stream://%(10000,0,350,440)') end Thanks everyone :D Diego On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: > I think its con.fileno in this case? ?Not sure. > /b > On May 3, 2009, at 4:00 PM, Diego Viola wrote: > > Yep, it works Guido. > > require 'socket' > > server = TCPServer.new(8084) > loop do > ???????con = server.accept > ???????con.puts "connect\n\n" > ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > answer\n\n" > ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" > end > > Thanks for the tip =D > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Sun May 3 14:33:07 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:33:07 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> Message-ID: <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> Will post some examples on the wiki now :) Diego On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: > NICE! It works, it works =D > > require 'socket' > require 'ESL' > > server = TCPServer.new(8084) > loop do > con = server.accept > fd = con.to_i > esl = ESL::ESLconnection.new(fd) > esl.execute('answer') > esl.execute('playback', 'tone_stream://%(10000,0,350,440)') > end > > Thanks everyone :D > > Diego > > On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >> I think its con.fileno in this case? ?Not sure. >> /b >> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >> >> Yep, it works Guido. >> >> require 'socket' >> >> server = TCPServer.new(8084) >> loop do >> ???????con = server.accept >> ???????con.puts "connect\n\n" >> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >> answer\n\n" >> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >> end >> >> Thanks for the tip =D >> >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! ?http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From diego.viola at gmail.com Sun May 3 14:43:30 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 3 May 2009 17:43:30 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> Message-ID: <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example Added. On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: > Will post some examples on the wiki now :) > > Diego > > On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: >> NICE! It works, it works =D >> >> require 'socket' >> require 'ESL' >> >> server = TCPServer.new(8084) >> loop do >> con = server.accept >> fd = con.to_i >> esl = ESL::ESLconnection.new(fd) >> esl.execute('answer') >> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') >> end >> >> Thanks everyone :D >> >> Diego >> >> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >>> I think its con.fileno in this case? ?Not sure. >>> /b >>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >>> >>> Yep, it works Guido. >>> >>> require 'socket' >>> >>> server = TCPServer.new(8084) >>> loop do >>> ???????con = server.accept >>> ???????con.puts "connect\n\n" >>> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>> answer\n\n" >>> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >>> end >>> >>> Thanks for the tip =D >>> >>> Brian West >>> brian at freeswitch.org >>> -- Meet us at ClueCon! ?http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > From dujinfang at gmail.com Sun May 3 21:01:42 2009 From: dujinfang at gmail.com (seven) Date: Mon, 4 May 2009 12:01:42 +0800 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <1240993632.22673.36.camel@localhost.localdomain> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> Message-ID: <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> Actually, for the "call back" agents, because the fifo use originate to start a new session, the new session won't hang up unless one agent answered or timeout. Agents will hear nothing and wait(member_wait=wait) on the queue or hanup(nowait) if caller hang up before an agent answer the phone. ' And I also found out the the member timeout doesn't work but call_timeout works in a dial string. Is it a bug I should reported to jira? {call_timeout=6,fifo_member_wait=nowait}user/1009@$${domain} And even the timeout works, it's not ideal. It's better to bridge to an agent other than originate I think. Keep looking. On Apr 29, 2009, at 4:27 PM, Fran?ois Delawarde wrote: > Hi, > > It should be easy to modify mod_fifo to include this functionality. > > Correct me if I'm wrong: > For "call back" agents at least, when X calls are in the the queue, > Freeswitch tries to search for up to X agents in database. This > algorithm is much more optimized than Asterisk, as Asterisk will > take calls one by one and try to connect them to an agent, it should > then stay as it is. > > The simplest idea to control the call distribution algorithm would > be to modify the database query in the "find_consumers" function > (right now, the algorithm is: "order by outbound_call_count"). A > variable could control the "order by" of this query, and the problem > would be solved at least for "call back" agents. I guess sqlite3 > should allow very complex queries, but I don't know if there could > be performance issues. > > Do you think it is a possible -trivial- solution? > > Fran?ois. > > On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: >> >> seven ha scritto: >> > oh, thank you Antonio. I think it would be better to collect more >> > ideas before open a bounty. And I more interested in >> playing(including >> > patching the code) with that than use the function. >> > >> I was working on other stuff yesterday and just looked at the wiki: >> - it seems there is already a bounty for something like that; >> - there is a wiki page about how to implement it with Javascript, ofc >> you need to tailor it to your own needs; >> >> AgX >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/93b2feef/attachment.html From codecomplete at free.fr Mon May 4 00:57:59 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 4 May 2009 00:57:59 -0700 (PDT) Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> Message-ID: <23364535.post@talk.nabble.com> Mitch Capper wrote: > You may want to look at the Intel Atom combo machines you can get a 1.6 > ghz machine probably for around $100-150 USD in a very small form factor > and very powerful. Thanks for the tip. Any link where I could check this out? The cheapest PC's I find are > 230? (Asus' EeePC), and with not enough room to stick a PCI card. Considering a Gigabyte GA-GC220 mobo sells for less than $50 retail + $60 for a PicoPSU, I'm surprised no one has come up with a mass-produced, CF-based $99 computer box :-/ That would be ideal to build a no-frill, virus-safe Linux box for web surfing... and a SOHO Freeswitch server. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23364535.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gallo at mctelefonia.com Mon May 4 01:19:31 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Mon, 04 May 2009 10:19:31 +0200 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <23364535.post@talk.nabble.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <23364535.post@talk.nabble.com> Message-ID: <49FEA513.8020109@mctelefonia.com> > The cheapest PC's I find are > 230? (Asus' EeePC), and with not enough room > to stick a PCI card. > Well actually a barebone with VIA C7 + 1 slot for PCI card (carefull you have to remove some metal part otherwise Sangoma/Digium card will not fit *LOL*) and external PSU are around 190/215? for 1 pieces. If 800/1200 Mhz are enough for you its a good choice. This hardware has problem with old kernel (LAN freezes). For even smaller hardware you can check "Acrosser" that AFAIK produces motherboard for poker/casino game machine. Or check the super small ALIX motherboard (those one has MINI-PCI) and fit in something like a router-metal-box and you'll stay around 95/120?. but with like 400 Mhz i think. > Considering a Gigabyte GA-GC220 mobo sells for less than $50 retail + $60 > for a PicoPSU, I'm surprised no one has come up with a mass-produced, > CF-based $99 computer box :-/ That would be ideal to build a no-frill, > virus-safe Linux box for web surfing... and a SOHO Freeswitch server. > Actually i dislike solution with only 1 internal PSU. As customer i will never purchase it because i know that at one point in time it will broke and i'll have the machine stopped for X hours of manteinances. With external PSU you can just plug in the new one. With 2 PSU you just need headspeakers to limit the soound from the PSU alarm noise :-P Also don't forget that if you're in europe and buy components and assemble them in a new product you have to make the "C.E." tests yourself (around 500/4000?) Antonio Gallo (agx) From codecomplete at free.fr Mon May 4 04:12:30 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 4 May 2009 04:12:30 -0700 (PDT) Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <49FEA513.8020109@mctelefonia.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <23364535.post@talk.nabble.com> <49FEA513.8020109@mctelefonia.com> Message-ID: <23366596.post@talk.nabble.com> Thanks Antonio for the links on Acrosser and PCEngines. It seems like PCE's alix1d is a good solution, provided 256MB is enough to hold Linux + Freeswitch + some tiny LAMP stack. Still, it looks like an Atom-included mobo like those from Asus or Gigabyte would be cheaper. The biggest issue is finding a case that allows for a PCI card + riser adapter that doesn't cost more than the mobo :-) -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23366596.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gallo at mctelefonia.com Mon May 4 05:05:57 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Mon, 04 May 2009 14:05:57 +0200 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <23366596.post@talk.nabble.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <23364535.post@talk.nabble.com> <49FEA513.8020109@mctelefonia.com> <23366596.post@talk.nabble.com> Message-ID: <49FEDA25.2050703@mctelefonia.com> Fred-145 ha scritto: > Thanks Antonio for the links on Acrosser and PCEngines. It seems like PCE's > alix1d is a good solution, provided 256MB is enough to hold Linux + > Freeswitch + some tiny LAMP stack. Still, it looks like an Atom-included > mobo like those from Asus or Gigabyte would be cheaper. The biggest issue is > finding a case that allows for a PCI card + riser adapter that doesn't cost > more than the mobo :-) > > Alix cases are like 6/9 ? from their shop site. I think its easy to find someone who work with aluminium that can make for you custom boxes for like like 6/20 ? at pcs. Since for them its just a fact of how much material is needed into the working process. If you cannot find any of this i know one that can make them in Bologna - Italy. Antonio Gallo (agx) From anthony.minessale at gmail.com Mon May 4 06:25:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 4 May 2009 08:25:04 -0500 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> Message-ID: <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> On Sun, May 3, 2009 at 11:01 PM, seven wrote: > Actually, for the "call back" agents, because the fifo use originate to > start a new session, the new session won't hang up unless one agent answered > or timeout. Agents will hear nothing and wait(member_wait=wait) on the > queue or hanup(nowait) if caller hang up before an agent answer the phone. ' > When you are using on-hook agents, it's presumed to be under low call volume, you can just set the agents to get popped into the queue in nowait mode so if the caller changed his mind the agent will get a hangup. Remember, if there are X customers in the queue, mod_fifo generates X outbound calls to try to service them. > > And I also found out the the member timeout doesn't work but call_timeout > works in a dial string. Is it a bug I should reported to jira? > > > lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@ > $${domain} > > call_timeout is only valid on inbound legs to set the timeout it's willing to wait for a caller to answer. You are confusing it with leg_timeout which is designed to go in the {} > > And even the timeout works, it's not ideal. It's better to bridge to an > agent other than originate I think. Keep looking. > I am not sure what you mean by that. bridge instead of originate? The process is to originate the call and then bridge the agent to the caller. All calls in FS start out as origiante???? If you want app_queue you are welcome to download and use it from http://www.asterisk.org > > On Apr 29, 2009, at 4:27 PM, Fran?ois Delawarde wrote: > > Hi, > > It should be easy to modify mod_fifo to include this functionality. > > Correct me if I'm wrong: > For "call back" agents at least, when X calls are in the the queue, > Freeswitch tries to search for up to X agents in database. This algorithm is > much more optimized than Asterisk, as Asterisk will take calls one by one > and try to connect them to an agent, it should then stay as it is. > > The simplest idea to control the call distribution algorithm would be to > modify the database query in the "find_consumers" function (right now, the > algorithm is: "order by outbound_call_count"). A variable could control the > "order by" of this query, and the problem would be solved at least for "call > back" agents. I guess sqlite3 should allow very complex queries, but I don't > know if there could be performance issues. > > Do you think it is a possible -trivial- solution? > > Fran?ois. > > On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: > > seven ha scritto: > > oh, thank you Antonio. I think it would be better to collect more > > ideas before open a bounty. And I more interested in playing(including > > patching the code) with that than use the function. > > > I was working on other stuff yesterday and just looked at the wiki: > - it seems there is already a bounty for something like that; > - there is a wiki page about how to implement it with Javascript, ofc > you need to tailor it to your own needs; > > AgX > > > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/39d83b5a/attachment.html From fdelawarde at wirelessmundi.com Mon May 4 07:55:10 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 04 May 2009 16:55:10 +0200 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> Message-ID: <1241448910.3016.127.camel@localhost.localdomain> Hello, Anthony, I would like to provide a patch allowing having different call distribution strategies, at least for "call back" agents. Do you think the simple approach of modifying the SQL query in find_consumers (given strategy that would be set from dialplan) would be enough? Thanks, Fran?ois. On Mon, 2009-05-04 at 08:25 -0500, Anthony Minessale wrote: > > > > On Sun, May 3, 2009 at 11:01 PM, seven wrote: > > Actually, for the "call back" agents, because the fifo use > originate to start a new session, the new session won't hang > up unless one agent answered or timeout. Agents will hear > nothing and wait(member_wait=wait) on the queue or > hanup(nowait) if caller hang up before an agent answer the > phone. ' > > > > When you are using on-hook agents, it's presumed to be under low call > volume, you can just set the agents to get popped > into the queue in nowait mode so if the caller changed his mind the > agent will get a hangup. Remember, if there are X customers in the > queue, mod_fifo generates X outbound calls to try to service them. > > > > > > And I also found out the the member timeout doesn't work but > call_timeout works in a dial string. Is it a bug I should > reported to jira? > > > > lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@ > $${domain} > > > > > call_timeout is only valid on inbound legs to set the timeout it's > willing to wait for a caller to answer. You are confusing it with > leg_timeout which is designed to go in the {} > > > > > And even the timeout works, it's not ideal. It's better to > bridge to an agent other than originate I think. Keep looking. > > > > I am not sure what you mean by that. bridge instead of originate? > The process is to originate the call and then bridge the agent to the > caller. All calls in FS start out as origiante???? > > If you want app_queue you are welcome to download and use it from > http://www.asterisk.org > > > > > On Apr 29, 2009, at 4:27 PM, Fran?ois Delawarde wrote: > > > Hi, > > > > It should be easy to modify mod_fifo to include this > > functionality. > > > > Correct me if I'm wrong: > > For "call back" agents at least, when X calls are in the the > > queue, Freeswitch tries to search for up to X agents in > > database. This algorithm is much more optimized than > > Asterisk, as Asterisk will take calls one by one and try to > > connect them to an agent, it should then stay as it is. > > > > The simplest idea to control the call distribution algorithm > > would be to modify the database query in the > > "find_consumers" function (right now, the algorithm is: > > "order by outbound_call_count"). A variable could control > > the "order by" of this query, and the problem would be > > solved at least for "call back" agents. I guess sqlite3 > > should allow very complex queries, but I don't know if there > > could be performance issues. > > > > Do you think it is a possible -trivial- solution? > > > > Fran?ois. > > > > On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: > > > > > seven ha scritto: > > > > oh, thank you Antonio. I think it would be better to collect more > > > > ideas before open a bounty. And I more interested in playing(including > > > > patching the code) with that than use the function. > > > > > > > I was working on other stuff yesterday and just looked at the wiki: > > > - it seems there is already a bounty for something like that; > > > - there is a wiki page about how to implement it with Javascript, ofc > > > you need to tailor it to your own needs; > > > > > > AgX > > > > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/413e633e/attachment-0001.html From dujinfang at gmail.com Mon May 4 07:59:09 2009 From: dujinfang at gmail.com (dujinfang) Date: Mon, 4 May 2009 22:59:09 +0800 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> Message-ID: On May 4, 2009, at 9:25 PM, Anthony Minessale wrote: > When you are using on-hook agents, it's presumed to be under low > call volume, you can just set the agents to get popped > into the queue in nowait mode so if the caller changed his mind the > agent will get a hangup. Remember, if there are X customers in the > queue, mod_fifo generates X outbound calls to try to service them. > Thank you. I'm more clear about the logic. I read some code, but need to read more to totally understand it. Here are two problems: 1) Is it possible to present the original caller id of the customer to the agent? 2) If the agent already on a call(busy), it would still call the extension and sim UAs generally allows multiple calls comes in. Is there a way to limit an agent can answer only one call?(mod_limit seems only limit outbound calls) Or do we need other complicated logic the figure out is the callee is busy before call them(possible by query the core db)? > > call_timeout is only valid on inbound legs to set the timeout it's > willing to wait for a caller to answer. You are confusing it with > leg_timeout which is designed to go in the {} > So it is supposed to hang up the caller when call_timeout timeout? I see it's deprecated on wiki: Deprecated - Use originate_timeout or leg_timeout. Controls how long (in seconds) to ring the B leg of a call when using the bridge application. But what i'm actually confusing is " I am having a problem with getting multiple Polycom IP phones to register to my Freeswitch server. Here is my setup (IP addresses are not actual ones, but are consistent throughout): Freeswitch server in colo facility IP addr: 1.1.1.1 (publicly routable) Linux NAT firewall router with iptables in office building external IP: 2.2.2.2 (publicly routable) internal IP: 192.168.1.1 (internal only, not publicly routable) Polycom IP301 phone A extension: 1001 IP addr: 192.168.1.2 Polycom IP301 phone B extension: 1002 IP addr: 192.168.1.3 snom 320 phone C extension: 1003 IP addr: 192.168.1.4 The Freeswitch server configuration has not changed much from the default installation. I tried changing NDLB-received-in-nat-reg-contact and it doesn't make a difference (although the register line adds a ";received=:" tag). Here is what happens: Polycom phone A registers successfully. If I execute "sofia status profile internal", I see this: Call-ID: f905cac7-125f0b1d-87aff436 at 192.168.1.2 User: 1001 at 1.1.1.1 Contact: "user" Agent: PolycomSoundPointIP-SPIP_301-UA/2.1.0.2708 Status: Registered(UDP-NAT)(unknown) EXP(2009-05-04 13:06:47) Host: 1.1.1.1 IP: 2.2.2.2 Port: 5060 Auth-User: 1001 Auth-Realm: 1.1.1.1 When Polycom phone B attempts to register, it cannot and I get the hollowed out phone icon on the phone display. I took a Wireshark capture and discovered that phone B does communicate with Freeswitch, but it is getting denied access. First phone B sends a REGISTER request: No. Time Source Destination Protocol Info 15114 117.617280 2.2.2.2 1.1.1.1 SIP Request: REGISTER sip:1.1.1.1:5060 Frame 15114 (561 bytes on wire, 561 bytes captured) Arrival Time: May 3, 2009 16:46:45.728592000 [Time delta from previous captured frame: 0.007070000 seconds] [Time delta from previous displayed frame: 10.505373000 seconds] [Time since reference or first frame: 117.617280000 seconds] Frame Number: 15114 Frame Length: 561 bytes Capture Length: 561 bytes [Frame is marked: False] [Protocols in frame: eth:ip:udp:sip] [Coloring Rule Name: UDP] [Coloring Rule String: udp] Ethernet II, Src: Xensourc_55:2a:dd (00:16:3e:55:2a:dd), Dst: D-Link_61:f2:9a (00:11:95:61:f2:9a) Destination: D-Link_61:f2:9a (00:11:95:61:f2:9a) Address: D-Link_61:f2:9a (00:11:95:61:f2:9a) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Source: Xensourc_55:2a:dd (00:16:3e:55:2a:dd) Address: Xensourc_55:2a:dd (00:16:3e:55:2a:dd) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Type: IP (0x0800) Internet Protocol, Src: 2.2.2.2 (2.2.2.2), Dst: 1.1.1.1 (1.1.1.1) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb0 (DSCP 0x2c: Unknown DSCP; ECN: 0x00) 1011 00.. = Differentiated Services Codepoint: Unknown (0x2c) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 547 Identification: 0x02ae (686) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 63 Protocol: UDP (0x11) Header checksum: 0x3ff3 [correct] [Good: True] [Bad : False] Source: 2.2.2.2 (2.2.2.2) Destination: 1.1.1.1 (1.1.1.1) User Datagram Protocol, Src Port: qsm-proxy (1164), Dst Port: sip (5060) Source port: qsm-proxy (1164) Destination port: sip (5060) Length: 527 Checksum: 0xdb1b [correct] [Good Checksum: True] [Bad Checksum: False] Session Initiation Protocol Request-Line: REGISTER sip:1.1.1.1:5060 SIP/2.0 Method: REGISTER [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK73c1d1c2BF65B36B Transport: UDP Sent-by Address: 192.168.1.3 Branch: z9hG4bK73c1d1c2BF65B36B From: "Rahim Orazkuliyev" ;tag=807917B4-BA73B497 SIP Display info: "Rahim Orazkuliyev" SIP from address: sip:1002 at 1.1.1.1 SIP tag: 807917B4-BA73B497 To: SIP to address: sip:1002 at 1.1.1.1 CSeq: 1 REGISTER Sequence Number: 1 Method: REGISTER Call-ID: 76909a58-dd169e7e-a7c5da19 at 192.168.1.3 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" Contact Binding: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" URI: SIP contact address: sip:1002 at 192.168.1.3 User-Agent: PolycomSoundPointIP-SPIP_301-UA/2.1.0.2708 Max-Forwards: 70 Expires: 3600 Content-Length: 0 The Freeswitch server responds as follows: No. Time Source Destination Protocol Info 15117 117.655406 1.1.1.1 2.2.2.2 SIP Status: 401 Unauthorized (0 bindings) Frame 15117 (698 bytes on wire, 698 bytes captured) Arrival Time: May 3, 2009 16:46:45.766718000 [Time delta from previous captured frame: 0.023210000 seconds] [Time delta from previous displayed frame: 0.038126000 seconds] [Time since reference or first frame: 117.655406000 seconds] Frame Number: 15117 Frame Length: 698 bytes Capture Length: 698 bytes [Frame is marked: False] [Protocols in frame: eth:ip:udp:sip] [Coloring Rule Name: UDP] [Coloring Rule String: udp] Ethernet II, Src: D-Link_61:f2:9a (00:11:95:61:f2:9a), Dst: Xensourc_55:2a:dd (00:16:3e:55:2a:dd) Destination: Xensourc_55:2a:dd (00:16:3e:55:2a:dd) Address: Xensourc_55:2a:dd (00:16:3e:55:2a:dd) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Source: D-Link_61:f2:9a (00:11:95:61:f2:9a) Address: D-Link_61:f2:9a (00:11:95:61:f2:9a) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Type: IP (0x0800) Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00) 1011 10.. = Differentiated Services Codepoint: Expedited Forwarding (0x2e) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 684 Identification: 0xd2e8 (53992) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 56 Protocol: UDP (0x11) Header checksum: 0x7627 [correct] [Good: True] [Bad : False] Source: 1.1.1.1 (1.1.1.1) Destination: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 664 Checksum: 0x1fb5 [correct] [Good Checksum: True] [Bad Checksum: False] Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK73c1d1c2BF65B36B;received=2.2.2.2 Transport: UDP Sent-by Address: 192.168.1.3 Branch: z9hG4bK73c1d1c2BF65B36B Received: 2.2.2.2 From: "Rahim Orazkuliyev" ;tag=807917B4-BA73B497 SIP Display info: "Rahim Orazkuliyev" SIP from address: sip:1002 at 1.1.1.1 SIP tag: 807917B4-BA73B497 To: ;tag=FXNpXtFFBBSpF SIP to address: sip:1002 at 1.1.1.1 SIP tag: FXNpXtFFBBSpF Call-ID: 76909a58-dd169e7e-a7c5da19 at 192.168.1.3 CSeq: 1 REGISTER Sequence Number: 1 Method: REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.4pre4-hacked Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="1.1.1.1", nonce="066baa69-aebb-4a39-a972-6d20625a79e0", algorithm=MD5, qop="auth" Authentication Scheme: Digest Realm: "1.1.1.1" Nonce Value: "066baa69-aebb-4a39-a972-6d20625a79e0" Algorithm: MD5 QOP: "auth" Content-Length: 0 Interestingly, the snom 320 phone registers fine: Call-ID: 3c26702fbd8b-3wozzv3bd908 User: 1003 at 1.1.1.1 Contact: "Wellie Chao" Agent: snom320/7.3.14 Status: Registered(UDP-NAT)(unknown) EXP(2009-05-04 14:32:50) Host: 1.1.1.1 IP: 2.2.2.2 Port: 2058 Auth-User: 1003 Auth-Realm: 1.1.1.1 I suspect that the snom 320 phone is working fine because the port is not 5060. The first Polycom (phone A) registered from port 5060. It appears that Freeswitch thinks the second Polycom (phone B) also is coming from port 5060 and is getting confused, thinking that phone B is trying to hijack phone A's registration. The packet capture was taken when running Freeswitch 1.0.4pre4, but I subsequently upgraded to 1.0.4pre6 and it didn't make a difference. The Linux NAT firewall router is running CentOS 5.3 with the most recent updates, and I have tried with ip_nat_sip and ip_conntrack_sip turned on and turned off. When ip_nat_sip and ip_conntrack_sip are turned on, I have included the 4 iptables rules needed: iptables -t nat -A POSTROUTING -o eth0 -j SNAT --to-source 2.2.2.2 iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT iptables -A INPUT -p udp --dport 5060 -j ACCEPT iptables -A FORWARD -o eth0 -p udp --dport 5060 -j ACCEPT It didn't make a difference no matter what I did. I am not sure why Freeswitch thinks that the source port is 5060 when it appears from the packet capture that the source port is 1164. Does anyone have any insights into why this is happening and what I can try to fix the problem? Regards, Wellie From mszlazak at aol.com Mon May 4 10:00:44 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 04 May 2009 13:00:44 -0400 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com> <069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org> Message-ID: <8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com> BTW Brian, Here is something that would make FS's VAD much better. The technique also improved Sphinx-3 performance in low-SNR enviroments and made it run over 40% faster. http://figment.cse.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Sat, 2 May 2009 7:42 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... On May 1, 2009, at 6:03 PM, mszlazak at aol.com wrote: Hi Moiz, I've checking out mod_pocketshinx against other asr's on Windows with the same hardware.? No matter what settings one tries, mod_pocketsphinx is virtually unusable in real world scenarios.? I have used it and it works fine... I think your expectations are a bit high for it... Complex things like dictation is not what PocketSphinx is for. ?You should try linux cuz I know it works great there. One can play around with mod_pocketsphinx settings so that it picks voice up well but then there better not be any background noise either from a bad connection or just everyday sounds.? There is no other ASR out there that doesn't get pissed off at background noise or any noise for that matter... have you called AT&T and Sprint lately? ?My dogs barking in the background really send theirs into fits and they paid tons of money for it. ? It just way to sensitive and of couse you'll notice this problem most with cell phones. Same with commercial ASR, Granted the acoustical model for PocketSphinx wasn't done with any files recorded from cellphone from what I can tell. ?You can do adaptation of the acoustical model as per the Sphinx wiki to make it more accurate for your needs.... that takes time and effort but it works. If you adjust the settings to try blocking out background noise you'll find you don't suceed all that well and then there are problems picking up the callers voice. Those settings are for telling when the person stopped talking... nothing more. It looks like some kind of signal pre-processing is required that isn't in place yet but we all know that this is a work-in progress. I'm not working on it... I run the pizza demo with PS and it works from my polycom rather well I would say it gets some things wrong but it does score them low so you can verify it in your scripts. I don't know if esl would make any difference. To use FS and an ASR/TTS I just bridge calls to another ASR application for now.? Mark Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/94029719/attachment-0001.html From brian at freeswitch.org Mon May 4 10:29:19 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 4 May 2009 12:29:19 -0500 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com> <069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org> <8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com> Message-ID: <00C14691-0334-4789-96C3-1A2D5F98CAD6@freeswitch.org> Wasn't aware Sphinx 3 was integrated into FreeSWITCH ... /b On May 4, 2009, at 12:00 PM, mszlazak at aol.com wrote: > BTW Brian, > > Here is something that would make FS's VAD much better. The > technique also improved Sphinx-3 performance in low-SNR enviroments > and made it run over 40% faster. > > http://figment.cse.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf > > Mark. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/c6dff190/attachment.html From mszlazak at aol.com Mon May 4 10:46:03 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 04 May 2009 13:46:03 -0400 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <00C14691-0334-4789-96C3-1A2D5F98CAD6@freeswitch.org> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com><069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org><8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com> <00C14691-0334-4789-96C3-1A2D5F98CAD6@freeswitch.org> Message-ID: <8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com> Nope it isn't but does that make a difference if pocketsphinx could use a similar upgrade? Anyway, you now have a way to make VAD better in FS. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Mon, 4 May 2009 10:29 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... Wasn't aware Sphinx 3 was integrated into FreeSWITCH ...? /b On May 4, 2009, at 12:00 PM, mszlazak at aol.com wrote: BTW Brian, Here is something that would make FS's VAD much better. The technique also improved Sphinx-3 performance in low-SNR enviroments and made it run over 40% faster. http://figment.cse.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf Mark. Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/189bf7fb/attachment.html From brian at freeswitch.org Mon May 4 10:56:26 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 4 May 2009 12:56:26 -0500 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: <8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com> References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com><069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org><8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com> <00C14691-0334-4789-96C3-1A2D5F98CAD6@freeswitch.org> <8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com> Message-ID: VAD isn't really high on my list right now. /b On May 4, 2009, at 12:46 PM, mszlazak at aol.com wrote: > Nope it isn't but does that make a difference if pocketsphinx could > use a similar upgrade? > Anyway, you now have a way to make VAD better in FS. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/72cf33a3/attachment.html From jalsot at gmail.com Mon May 4 10:59:13 2009 From: jalsot at gmail.com (Tamas) Date: Mon, 04 May 2009 19:59:13 +0200 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com><069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org><8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com> <00C14691-0334-4789-96C3-1A2D5F98CAD6@freeswitch.org> <8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com> Message-ID: <49FF2CF1.8050905@gmail.com> Hi, maybe things in speex should be worth to look for too. Just my 2 cents... Regards, Tamas ps: It seems, VAD+DTX in mod_speex does not work. Brian West ?rta: > VAD isn't really high on my list right now. > > /b > > On May 4, 2009, at 12:46 PM, mszlazak at aol.com > wrote: > >> Nope it isn't but does that make a difference if pocketsphinx could >> use a similar upgrade? >> Anyway, you now have a way to make VAD better in FS. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Mon May 4 11:10:12 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 04 May 2009 14:10:12 -0400 Subject: [Freeswitch-users] Ways of Integrating Sphinx... In-Reply-To: References: <29b888f80905010638t20bbc640wd01ae6dc1bec033f@mail.gmail.com><8CB98C82A4A45AF-F54-56D@webmail-dx21.sysops.aol.com><069F7705-86A2-4D8B-AEED-1EB5D71A5328@freeswitch.org><8CB9AF0FF5E1CEF-1014-9E7@WEBMAIL-DY21.sysops.aol.com><00C14691-0334-4789-96C3-1A2D5F98CAD6@freeswitch.org><8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com> Message-ID: <8CB9AFAB3BAF4CD-1E8C-11A6@WEBMAIL-DY08.sysops.aol.com> No problem but at least this reference could be used in the future. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Mon, 4 May 2009 10:56 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... VAD isn't really high on my list right now. /b On May 4, 2009, at 12:46 PM, mszlazak at aol.com wrote: Nope it isn't but does that make a difference if pocketsphinx could use a similar upgrade? Anyway, you now have a way to make VAD better in FS. Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/d9c0dbb3/attachment-0001.html From austad at signal15.com Mon May 4 11:52:17 2009 From: austad at signal15.com (Jay Austad) Date: Mon, 4 May 2009 13:52:17 -0500 Subject: [Freeswitch-users] XML editors and freeswitch XML spec compliance? Message-ID: <9624673E-4825-4218-91AD-133527B43FF1@signal15.com> Has anyone found a decent XML editor for the XML files? I loaded the files up in OrangeVolt under Eclipse, but didn't have much time to play around with it. Also, I tried opening them with an XML plugin for TextMate, and it says the files are not compliant. The reason is that the first line of each files is a comment describing the file, followed by the tag. Apparently, the tag needs to be the first line in the file for compliance to the spec. -- jay austad | 612.423.1433 | austad at signal15.com From brian at freeswitch.org Mon May 4 12:05:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 4 May 2009 14:05:18 -0500 Subject: [Freeswitch-users] XML editors and freeswitch XML spec compliance? In-Reply-To: <9624673E-4825-4218-91AD-133527B43FF1@signal15.com> References: <9624673E-4825-4218-91AD-133527B43FF1@signal15.com> Message-ID: <0143E14B-9599-4C68-9604-F4F75E89713E@freeswitch.org> On May 4, 2009, at 1:52 PM, Jay Austad wrote: > Has anyone found a decent XML editor for the XML files? I loaded the > files up in OrangeVolt under Eclipse, but didn't have much time to > play around with it. > > Also, I tried opening them with an XML plugin for TextMate, and it > says the files are not compliant. The reason is that the first line > of each files is a comment describing the file, followed by the > tag. Apparently, the tag needs to be the first line in the file > for compliance to the spec. I have moved the one file a few weeks back that had the tag (freeswitch.xml) in it to the top of the file.. the others are preprocessed and included by FreeSWITCH which doesn't care about the tag itself. The files can be combined into a large freeswitch.xml without any includes! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/5e5dc83a/attachment.html From msc at freeswitch.org Mon May 4 12:12:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 4 May 2009 12:12:00 -0700 Subject: [Freeswitch-users] XML editors and freeswitch XML spec compliance? In-Reply-To: <9624673E-4825-4218-91AD-133527B43FF1@signal15.com> References: <9624673E-4825-4218-91AD-133527B43FF1@signal15.com> Message-ID: <87f2f3b90905041212j1b868c2fwa0faab1ace355e08@mail.gmail.com> I recommend something that does syntax highlighting without all the pedantic nonsense. In a Windows environment I've used notepad++ and notepad2. In a Linux environment I've just used Kedit in the GUI environment but emacs and vim work nicely in the text-only environment. In OSX I use emacs in a texty environment and good ol' TextMate (no bundle) in the GUI environment. You have many choices. If you need something more "powerful" then I would look to see if you can tell your editor not to get its knickers in a twist when "" isn't on the first line of the file. :) -MC On Mon, May 4, 2009 at 11:52 AM, Jay Austad wrote: > Has anyone found a decent XML editor for the XML files? I loaded the > files up in OrangeVolt under Eclipse, but didn't have much time to > play around with it. > > Also, I tried opening them with an XML plugin for TextMate, and it > says the files are not compliant. The reason is that the first line > of each files is a comment describing the file, followed by the > tag. Apparently, the tag needs to be the first line in the file > for compliance to the spec. > > -- > jay austad | 612.423.1433 | austad at signal15.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/3995e2f9/attachment.html From msc at freeswitch.org Mon May 4 13:50:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 4 May 2009 13:50:36 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH mod_opal Now Officially In Beta Message-ID: <87f2f3b90905041350q5c896820o537c17e35abd690@mail.gmail.com> The FreeSWITCH team would like everyone to know that the mod_opal module is now officially in beta. Please read this article for more information: http://www.freeswitch.org/node/179 Many thanks to Robert Jongbloed and Craig Southeren of the OPAL project for their many years of support for open source VoIP. -Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/6e290df2/attachment.html From Prometheus001 at gmx.net Mon May 4 16:09:58 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 05 May 2009 01:09:58 +0200 Subject: [Freeswitch-users] Invite on SIP instead of TLS Message-ID: <49FF75C6.9060501@gmx.net> I have updated my system from SVN 10003 to 13223. I also have updated all libraries etc. Everything works fine (SIP +TLS) when calling internal numbers (conference). However calling internally registered phones does not work. Here are some facts which do not fit together - phones are sucessfully registered via TLS (ok) - debug log show that phone will be called via TLS (port 5061) (ok) - Invite message however is sent via SIP (port 5060) Please see the part of the logs below. Anybody has a clue what happened here? Best regards Peter Debug Log: ============ 2009-05-05 00:49:38 [DEBUG] sofia_glue.c:1599 sofia_glue_do_invite() sip:723329 at 217.xxx.xxx.186:2651 Setting proxy route to sofia/internal/sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 2009-05-05 00:49:38 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651) State CONSUME_MEDIA 2009-05-05 00:49:38 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/internal/sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 entering state [calling][0] 2009-05-05 00:49:38 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/internal/sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 entering state [terminated][503] 2009-05-05 00:49:38 [NOTICE] sofia.c:3469 sofia_handle_sip_i_state() Hangup sofia/internal/sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da;fs_nat=yes;fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] Phone is registered with TLS ============================ Call-ID: OWE1NTAyNWU0MGE2NjI1OWJhZjM1YWJiYWJjZGYzYTI. User: 723329 at sip2.mydomain.de Contact: "723329" Agent: eyeBeam release 1102u stamp 52345 Status: Registered(TLS-NAT)(unknown) EXP(2009-05-05 00:58:03) Host: sip2.mydomain.de IP: 217.xxx.xxx.186 Port: 2651 Auth-User: 723329 Auth-Realm: sip2.mydomain.de SIP message instead of TLS message: ==================== U 217.xxx.xxx.190:5060 -> 217.xxx.xxx.186:2651 INVITE sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190;rport;branch=z9hG4bK2tH444a02mQZc. Route: . Max-Forwards: 69. From: "Extension 723321" ;tag=HB02U2mHX28yK. To: . Call-ID: e3b50c2c-b3a0-122c-4491-001e904cc34e. CSeq: 114620396 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13223M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 447. P-Key-Flags: keys="3". Remote-Party-ID: "Extension 723321" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1314931392159531063 5177685988992248857 IN IP4 217.xxx.xxx.190. s=FreeSWITCH. c=IN IP4 217.xxx.xxx.190. t=0 0. m=audio 12556 RTP/SAVP 8 9 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:w/ubmPAP4I5BA1Gv1ZWZzbJkfst2e4cY7bKedcjA. From brian at freeswitch.org Mon May 4 16:18:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 4 May 2009 18:18:11 -0500 Subject: [Freeswitch-users] Invite on SIP instead of TLS In-Reply-To: <49FF75C6.9060501@gmx.net> References: <49FF75C6.9060501@gmx.net> Message-ID: <91A1CA71-4F1C-4B46-9997-3F98159058A0@freeswitch.org> I'm pretty sure this was fixed in 13226 please update. You're using a new feature it seems. /b On May 4, 2009, at 6:09 PM, Peter P GMX wrote: > fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From dujinfang at gmail.com Mon May 4 20:50:13 2009 From: dujinfang at gmail.com (seven) Date: Tue, 5 May 2009 11:50:13 +0800 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> Message-ID: <89C98A29-AE97-4366-9729-0FC41FE8AD36@gmail.com> On May 4, 2009, at 9:25 PM, Anthony Minessale wrote: > > > On Sun, May 3, 2009 at 11:01 PM, seven wrote: > Actually, for the "call back" agents, because the fifo use originate > to start a new session, the new session won't hang up unless one > agent answered or timeout. Agents will hear nothing and > wait(member_wait=wait) on the queue or hanup(nowait) if caller hang > up before an agent answer the phone. ' > > > When you are using on-hook agents, it's presumed to be under low > call volume, you can just set the agents to get popped > into the queue in nowait mode so if the caller changed his mind the > agent will get a hangup. Remember, if there are X customers in the > queue, mod_fifo generates X outbound calls to try to service them. > Actually it generates N(=member count) outbound calls as the waiting > 0 before the customer be serviced(answered) by the agent in the on- hook mode. I might can make a patch but not sure if that affect the off-hook agents. 2009-05-05 11:42:44 [INFO] mod_fifo.c:574 node_thread_run() sales_fifo at 192.168.1.27 waiting 1 consumer_total 0 idle_consumers 0 2009-05-05 11:42:45 [INFO] mod_fifo.c:574 node_thread_run() sales_fifo at 192.168.1.27 waiting 1 consumer_total 0 idle_consumers 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/b8cd351b/attachment-0001.html From dujinfang at gmail.com Mon May 4 20:53:49 2009 From: dujinfang at gmail.com (seven) Date: Tue, 5 May 2009 11:53:49 +0800 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <1241448910.3016.127.camel@localhost.localdomain> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> <1241448910.3016.127.camel@localhost.localdomain> Message-ID: <2B9729E0-E301-4871-A349-ADF27ED83B4B@gmail.com> I think order by outbound_call_count will cause problem. Think about the fifo run a few days, and we added a new member, the outbound_call_count will always less than others in a certain time. What about use order by next_avail? On May 4, 2009, at 10:55 PM, Fran?ois Delawarde wrote: > Hello, > > Anthony, I would like to provide a patch allowing having different > call distribution strategies, at least for "call back" agents. > > Do you think the simple approach of modifying the SQL query in > find_consumers (given strategy that would be set from dialplan) > would be enough? > > Thanks, > Fran?ois. > > On Mon, 2009-05-04 at 08:25 -0500, Anthony Minessale wrote: >> >> >> On Sun, May 3, 2009 at 11:01 PM, seven wrote: >> Actually, for the "call back" agents, because the fifo use >> originate to start a new session, the new session won't hang up >> unless one agent answered or timeout. Agents will hear nothing and >> wait(member_wait=wait) on the queue or hanup(nowait) if caller hang >> up before an agent answer the phone. ' >> >> >> >> When you are using on-hook agents, it's presumed to be under low >> call volume, you can just set the agents to get popped >> into the queue in nowait mode so if the caller changed his mind the >> agent will get a hangup. Remember, if there are X customers in the >> queue, mod_fifo generates X outbound calls to try to service them. >> >> >> >> >> >> And I also found out the the member timeout doesn't work but >> call_timeout works in a dial string. Is it a bug I should reported >> to jira? >> >> >> >> > lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$$ >> {domain} >> >> >> >> >> call_timeout is only valid on inbound legs to set the timeout it's >> willing to wait for a caller to answer. You are confusing it with >> leg_timeout which is designed to go in the {} >> >> >> >> >> And even the timeout works, it's not ideal. It's better to bridge >> to an agent other than originate I think. Keep looking. >> >> >> >> I am not sure what you mean by that. bridge instead of originate? >> The process is to originate the call and then bridge the agent to >> the caller. All calls in FS start out as origiante???? >> >> If you want app_queue you are welcome to download and use it from http://www.asterisk.org >> >> >> >> On Apr 29, 2009, at 4:27 PM, Fran?ois Delawarde wrote: >>> Hi, >>> >>> It should be easy to modify mod_fifo to include this functionality. >>> >>> Correct me if I'm wrong: >>> For "call back" agents at least, when X calls are in the the >>> queue, Freeswitch tries to search for up to X agents in database. >>> This algorithm is much more optimized than Asterisk, as Asterisk >>> will take calls one by one and try to connect them to an agent, it >>> should then stay as it is. >>> >>> The simplest idea to control the call distribution algorithm would >>> be to modify the database query in the "find_consumers" function >>> (right now, the algorithm is: "order by outbound_call_count"). A >>> variable could control the "order by" of this query, and the >>> problem would be solved at least for "call back" agents. I guess >>> sqlite3 should allow very complex queries, but I don't know if >>> there could be performance issues. >>> >>> Do you think it is a possible -trivial- solution? >>> >>> Fran?ois. >>> >>> On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: >>>> >>>> seven ha scritto: >>>> > oh, thank you Antonio. I think it would be better to collect more >>>> > ideas before open a bounty. And I more interested in >>>> playing(including >>>> > patching the code) with that than use the function. >>>> > >>>> I was working on other stuff yesterday and just looked at the wiki: >>>> - it seems there is already a bounty for something like that; >>>> - there is a wiki page about how to implement it with Javascript, >>>> ofc >>>> you need to tailor it to your own needs; >>>> >>>> AgX >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/45c6645d/attachment.html From stevecrozz at gmail.com Mon May 4 22:12:49 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 4 May 2009 22:12:49 -0700 Subject: [Freeswitch-users] help with mod_conference stability Message-ID: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> We had our first big issues with our freeswitch system today. During at least 2 conferences, audio became jittery and there were three occasions where everyone was dropped from a conference. Even so, conference recording was not interrupted, and the freeswitch debug log doesn't show anything unusual. Our hardware monitoring software doesn't show any kind of unusual resources usage, and our web host claims there were no outages during the time when we were experiencing problems. We're currently running revision 12259. How should I proceed in diagnosing this issue? --Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/b3c268b5/attachment.html From jason at jasonjgw.net Mon May 4 22:27:48 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 5 May 2009 15:27:48 +1000 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> Message-ID: <20090505052748.GA8939@jdc.jasonjgw.net> Stephen Crosby wrote: > We had our first big issues with our freeswitch system today. During at > least 2 conferences, audio became jittery and there were three occasions > where everyone was dropped from a conference. Even so, conference recording > was not interrupted, and the freeswitch debug log doesn't show anything > unusual. My suspicion is that it's a network problem unrelated to FreeSWITCH. Did you try pinging the host while the problems were occurring? What did FreeSWITCH report in the log as the reason for terminating the calls to the conference? From stevecrozz at gmail.com Mon May 4 22:52:17 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 4 May 2009 22:52:17 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <20090505052748.GA8939@jdc.jasonjgw.net> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> Message-ID: <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> Network problem is what I'm still thinking. Take a look at this log snippet: http://pastebin.freeswitch.org/8813 I'm CALLER_A, and you can see me calling back in on line 8 after I got dropped. But that was only seconds prior. There really seems to be nothing in the log at the time the calls were dropped. --Stephen On Mon, May 4, 2009 at 10:27 PM, Jason White wrote: > Stephen Crosby wrote: > > We had our first big issues with our freeswitch system today. During at > > least 2 conferences, audio became jittery and there were three occasions > > where everyone was dropped from a conference. Even so, conference > recording > > was not interrupted, and the freeswitch debug log doesn't show anything > > unusual. > > My suspicion is that it's a network problem unrelated to FreeSWITCH. > > Did you try pinging the host while the problems were occurring? > > What did FreeSWITCH report in the log as the reason for terminating the > calls > to the conference? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/07e1812b/attachment-0001.html From stevecrozz at gmail.com Mon May 4 22:55:50 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 4 May 2009 22:55:50 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> Message-ID: <11990ade0905042255r4e46b590t5fe49068fefdf3f1@mail.gmail.com> To answer the other question, I did not try pinging the host while the problems were occurring, but I was able to call back in with no problem only a few seconds after I got dropped (less than 20), so if it is a network issue it would have to have been very brief. --Stephen On Mon, May 4, 2009 at 10:52 PM, Stephen Crosby wrote: > Network problem is what I'm still thinking. Take a look at this log > snippet: > http://pastebin.freeswitch.org/8813 > > I'm CALLER_A, and you can see me calling back in on line 8 after I got > dropped. But that was only seconds prior. There really seems to be nothing > in the log at the time the calls were dropped. > > --Stephen > > > On Mon, May 4, 2009 at 10:27 PM, Jason White wrote: > >> Stephen Crosby wrote: >> > We had our first big issues with our freeswitch system today. During at >> > least 2 conferences, audio became jittery and there were three occasions >> > where everyone was dropped from a conference. Even so, conference >> recording >> > was not interrupted, and the freeswitch debug log doesn't show anything >> > unusual. >> >> My suspicion is that it's a network problem unrelated to FreeSWITCH. >> >> Did you try pinging the host while the problems were occurring? >> >> What did FreeSWITCH report in the log as the reason for terminating the >> calls >> to the conference? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/020ce069/attachment.html From stevecrozz at gmail.com Mon May 4 23:09:05 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 4 May 2009 23:09:05 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042255r4e46b590t5fe49068fefdf3f1@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> <11990ade0905042255r4e46b590t5fe49068fefdf3f1@mail.gmail.com> Message-ID: <11990ade0905042309g402c3630lb492391643d580c2@mail.gmail.com> Sorry for the extra messages, but I've just discovered something that probably helps pinpoint the problem: http://pastebin.freeswitch.org/8814 It seems that the callers were disconnected, but freeswitch had to wait a timeout period before it actually hangs up which looks like about 5 minutes. So I was disconnected from a conference, then I called back in, then freeswitch later hung up on the first call. This seems very much like a network problem. What can I do to fix it? --Stephen On Mon, May 4, 2009 at 10:55 PM, Stephen Crosby wrote: > To answer the other question, I did not try pinging the host while the > problems were occurring, but I was able to call back in with no problem only > a few seconds after I got dropped (less than 20), so if it is a network > issue it would have to have been very brief. > > --Stephen > > > On Mon, May 4, 2009 at 10:52 PM, Stephen Crosby wrote: > >> Network problem is what I'm still thinking. Take a look at this log >> snippet: >> http://pastebin.freeswitch.org/8813 >> >> I'm CALLER_A, and you can see me calling back in on line 8 after I got >> dropped. But that was only seconds prior. There really seems to be nothing >> in the log at the time the calls were dropped. >> >> --Stephen >> >> >> On Mon, May 4, 2009 at 10:27 PM, Jason White wrote: >> >>> Stephen Crosby wrote: >>> > We had our first big issues with our freeswitch system today. During at >>> > least 2 conferences, audio became jittery and there were three >>> occasions >>> > where everyone was dropped from a conference. Even so, conference >>> recording >>> > was not interrupted, and the freeswitch debug log doesn't show anything >>> > unusual. >>> >>> My suspicion is that it's a network problem unrelated to FreeSWITCH. >>> >>> Did you try pinging the host while the problems were occurring? >>> >>> What did FreeSWITCH report in the log as the reason for terminating the >>> calls >>> to the conference? >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/0d0d0fa1/attachment.html From jason at jasonjgw.net Mon May 4 23:11:53 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 5 May 2009 16:11:53 +1000 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> Message-ID: <20090505061153.GA12919@jdc.jasonjgw.net> Stephen Crosby wrote: > Network problem is what I'm still thinking. Take a look at this log snippet: > > http://pastebin.freeswitch.org/8813 > > I'm CALLER_A, and you can see me calling back in on line 8 after I got > dropped. But that was only seconds prior. There really seems to be nothing > in the log at the time the calls were dropped. In particular, the "dropped" calls weren't shown to have terminated at that point. Can you search through the logs and find out when they did terminate, and why? A time-out, perhaps? Assuming that FreeSWITCH didn't crash, and show channels indicates that those sessions aren't still current, they must have timed out or otherwise terminated at some point, which would have left log messages. I've been in situations involving brief network outages (say, 10 seconds) after which FreeSWITCH conferences have resumed as normal, with no dropped connections. However, those circumstances involved FreeSWITCH instances on both ends of the call; it is possible, though uninformed speculation on my part, that some other devices might time out quicker in the event of network issues. Another test you could try, if this persists, is to run FreeSWITCH locally and use it to call the conference. If the call terminates abnormally, you will then have the logs at both ends. Of course, your phone might have logs itself, but the FreeSWITCH ones are probably better. Your report of jitter is also indicative of a less than reliable network. From jason at jasonjgw.net Mon May 4 23:22:00 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 5 May 2009 16:22:00 +1000 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042309g402c3630lb492391643d580c2@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> <11990ade0905042255r4e46b590t5fe49068fefdf3f1@mail.gmail.com> <11990ade0905042309g402c3630lb492391643d580c2@mail.gmail.com> Message-ID: <20090505062200.GC12919@jdc.jasonjgw.net> Stephen Crosby wrote: > Sorry for the extra messages, but I've just discovered something that > probably helps pinpoint the problem: > http://pastebin.freeswitch.org/8814 > > It seems that the callers were disconnected, but freeswitch had to wait a > timeout period before it actually hangs up which looks like about 5 minutes. > So I was disconnected from a conference, then I called back in, then > freeswitch later hung up on the first call. This seems very much like a > network problem. Yes, and it was the device at your end that terminated the call. I would start by setting up alarm pings (say, ping -a) on my local machine and listening for jitter and drop-outs. If it turns out to be a network problem, then it's very likely an issue between you and your hosting provider at that point. From stevecrozz at gmail.com Mon May 4 23:34:30 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Mon, 4 May 2009 23:34:30 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <20090505062200.GC12919@jdc.jasonjgw.net> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> <11990ade0905042255r4e46b590t5fe49068fefdf3f1@mail.gmail.com> <11990ade0905042309g402c3630lb492391643d580c2@mail.gmail.com> <20090505062200.GC12919@jdc.jasonjgw.net> Message-ID: <11990ade0905042334m267780a8y9d727f9938352774@mail.gmail.com> I was on this conference and there were 4 of us that got dropped at the same time. By the way, these were all regular PSTN lines calling through our SIP provider and being routed to our freeswitch instance which is being hosted on a VPS. When you say a device on my end terminated the call, do you mean our home telephones which we used to dial in or the freeswitch box? --Stephen On Mon, May 4, 2009 at 11:22 PM, Jason White wrote: > Stephen Crosby wrote: > > Sorry for the extra messages, but I've just discovered something that > > probably helps pinpoint the problem: > > http://pastebin.freeswitch.org/8814 > > > > It seems that the callers were disconnected, but freeswitch had to wait a > > timeout period before it actually hangs up which looks like about 5 > minutes. > > So I was disconnected from a conference, then I called back in, then > > freeswitch later hung up on the first call. This seems very much like a > > network problem. > > Yes, and it was the device at your end that terminated the call. > > I would start by setting up alarm pings (say, ping -a) on my local machine > and > listening for jitter and drop-outs. If it turns out to be a network > problem, > then it's very likely an issue between you and your hosting provider at > that > point. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090504/ca782e09/attachment.html From jason at jasonjgw.net Mon May 4 23:49:12 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 5 May 2009 16:49:12 +1000 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042334m267780a8y9d727f9938352774@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <20090505052748.GA8939@jdc.jasonjgw.net> <11990ade0905042252ofd36f23m5086ea3c003fcff2@mail.gmail.com> <11990ade0905042255r4e46b590t5fe49068fefdf3f1@mail.gmail.com> <11990ade0905042309g402c3630lb492391643d580c2@mail.gmail.com> <20090505062200.GC12919@jdc.jasonjgw.net> <11990ade0905042334m267780a8y9d727f9938352774@mail.gmail.com> Message-ID: <20090505064912.GA21875@jdc.jasonjgw.net> Stephen Crosby wrote: > I was on this conference and there were 4 of us that got dropped at the same > time. By the way, these were all regular PSTN lines calling through our SIP > provider and being routed to our freeswitch instance which is being hosted > on a VPS. > > When you say a device on my end terminated the call, do you mean our home > telephones which we used to dial in or the freeswitch box? No, I mean the SIP device managing your end of the connection, which in that case would be your SIP -> PSTN provider. From dujinfang at gmail.com Tue May 5 00:06:26 2009 From: dujinfang at gmail.com (seven) Date: Tue, 5 May 2009 15:06:26 +0800 Subject: [Freeswitch-users] any way ring fifo members one by one? In-Reply-To: <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> References: <7A2B3C96-207C-4EDF-A6B7-8EA17A4FC1E0@gmail.com> <191c3a030904280533k4ca3c41fy9bd58c5c137abd86@mail.gmail.com> <26102C50-1969-4D01-A255-E2530D37CC1E@gmail.com> <191c3a030904280724j68deb0b1k6d3afe5a63f9dd67@mail.gmail.com> <49F72337.9050602@mctelefonia.com> <75CEADE3-F516-4E9A-B860-3B7CAA6773FE@gmail.com> <49F7F7C0.4050908@mctelefonia.com> <1240993632.22673.36.camel@localhost.localdomain> <433C9410-8679-42ED-984F-F4BF694A10E6@gmail.com> <191c3a030905040625j7677fdd7kf200ac811f6a7794@mail.gmail.com> Message-ID: here is my patch: http://jira.freeswitch.org/browse/MODAPP-272 On May 4, 2009, at 9:25 PM, Anthony Minessale wrote: > > > On Sun, May 3, 2009 at 11:01 PM, seven wrote: > Actually, for the "call back" agents, because the fifo use originate > to start a new session, the new session won't hang up unless one > agent answered or timeout. Agents will hear nothing and > wait(member_wait=wait) on the queue or hanup(nowait) if caller hang > up before an agent answer the phone. ' > > > When you are using on-hook agents, it's presumed to be under low > call volume, you can just set the agents to get popped > into the queue in nowait mode so if the caller changed his mind the > agent will get a hangup. Remember, if there are X customers in the > queue, mod_fifo generates X outbound calls to try to service them. > > > > > And I also found out the the member timeout doesn't work but > call_timeout works in a dial string. Is it a bug I should reported > to jira? > > > lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$$ > {domain} > > > > call_timeout is only valid on inbound legs to set the timeout it's > willing to wait for a caller to answer. You are confusing it with > leg_timeout which is designed to go in the {} > > > > And even the timeout works, it's not ideal. It's better to bridge to > an agent other than originate I think. Keep looking. > > I am not sure what you mean by that. bridge instead of originate? > The process is to originate the call and then bridge the agent to > the caller. All calls in FS start out as origiante???? > > If you want app_queue you are welcome to download and use it from http://www.asterisk.org > > > > On Apr 29, 2009, at 4:27 PM, Fran?ois Delawarde wrote: >> Hi, >> >> It should be easy to modify mod_fifo to include this functionality. >> >> Correct me if I'm wrong: >> For "call back" agents at least, when X calls are in the the queue, >> Freeswitch tries to search for up to X agents in database. This >> algorithm is much more optimized than Asterisk, as Asterisk will >> take calls one by one and try to connect them to an agent, it >> should then stay as it is. >> >> The simplest idea to control the call distribution algorithm would >> be to modify the database query in the "find_consumers" function >> (right now, the algorithm is: "order by outbound_call_count"). A >> variable could control the "order by" of this query, and the >> problem would be solved at least for "call back" agents. I guess >> sqlite3 should allow very complex queries, but I don't know if >> there could be performance issues. >> >> Do you think it is a possible -trivial- solution? >> >> Fran?ois. >> >> On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: >>> >>> seven ha scritto: >>> > oh, thank you Antonio. I think it would be better to collect more >>> > ideas before open a bounty. And I more interested in >>> playing(including >>> > patching the code) with that than use the function. >>> > >>> I was working on other stuff yesterday and just looked at the wiki: >>> - it seems there is already a bounty for something like that; >>> - there is a wiki page about how to implement it with Javascript, >>> ofc >>> you need to tailor it to your own needs; >>> >>> AgX >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/a0723954/attachment.html From dujinfang at gmail.com Tue May 5 03:10:56 2009 From: dujinfang at gmail.com (seven) Date: Tue, 5 May 2009 18:10:56 +0800 Subject: [Freeswitch-users] Got more 404s than should. Message-ID: <85AEC36D-9D7D-4C47-B5BC-3A73B208EFA4@gmail.com> Hi, Please help me take a look into this: http://pastebin.freeswitch.org/8816 My problem is why the first one keep sending 404s even got the ACK? It seems that in the first test the ACK is not recognized by FS so it keep sending 404. The only difference I can see of the two ACK-404 is the first one missing a tag of TO, is that related? Shouldn't the tag optional? PS: it works OK if I dial an exists number and get answered. It only happens on unanswered messages like 404 or 503 etc. Thank you. From Prometheus001 at gmx.net Tue May 5 03:22:50 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 05 May 2009 12:22:50 +0200 Subject: [Freeswitch-users] "conf-is-unlocked.wav" missing Message-ID: <4A00137A.3070701@gmx.net> Hello, I tried conferencing for FS und tried to lock/unlock conferences. While "conf-is-locked.wav" was played, "conf-is-unlocked.wav" was missing in the file system. Any idea where I can download this? Best regards Peter From jason at jasonjgw.net Tue May 5 03:35:25 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 5 May 2009 20:35:25 +1000 Subject: [Freeswitch-users] "conf-is-unlocked.wav" missing In-Reply-To: <4A00137A.3070701@gmx.net> References: <4A00137A.3070701@gmx.net> Message-ID: <20090505103525.GA28313@jdc.jasonjgw.net> Peter P GMX wrote: > I tried conferencing for FS und tried to lock/unlock conferences. > While "conf-is-locked.wav" was played, "conf-is-unlocked.wav" was > missing in the file system. It seems to be missing from the FreeSWITCH sound files. I have all versions including 48k, and it isn't there. You could record a sound of your own, or just set is-unlocked-sound to refer to the same file as is-locked-sound. By default this is in conference.conf.xml. From brian at freeswitch.org Tue May 5 03:41:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 5 May 2009 05:41:02 -0500 Subject: [Freeswitch-users] Got more 404s than should. In-Reply-To: <85AEC36D-9D7D-4C47-B5BC-3A73B208EFA4@gmail.com> References: <85AEC36D-9D7D-4C47-B5BC-3A73B208EFA4@gmail.com> Message-ID: <34EA62EA-5133-43A9-AE2A-E8DFD055BEFB@freeswitch.org> It appears to be a broken client. Your client doesn't ack with the to tag like zoiper does. /b On May 5, 2009, at 5:10 AM, seven wrote: > My problem is why the first one keep sending 404s even got the ACK? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/11376dd0/attachment.html From brian at freeswitch.org Tue May 5 03:49:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 5 May 2009 05:49:34 -0500 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> Message-ID: <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> First off you're not on SVN trunk secondly Are you executing the conference app inside your js file? If so then there could be the problem! You have also forgotten to include anything about Distro, OS, CPU and Memory. /b On May 5, 2009, at 12:12 AM, Stephen Crosby wrote: > We had our first big issues with our freeswitch system today. During > at least 2 conferences, audio became jittery and there were three > occasions where everyone was dropped from a conference. Even so, > conference recording was not interrupted, and the freeswitch debug > log doesn't show anything unusual. > > Our hardware monitoring software doesn't show any kind of unusual > resources usage, and our web host claims there were no outages > during the time when we were experiencing problems. > > We're currently running revision 12259. > > How should I proceed in diagnosing this issue? > > --Stephen Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/22f776b0/attachment-0001.html From dujinfang at gmail.com Tue May 5 04:29:00 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 5 May 2009 19:29:00 +0800 Subject: [Freeswitch-users] Got more 404s than should. In-Reply-To: <34EA62EA-5133-43A9-AE2A-E8DFD055BEFB@freeswitch.org> References: <85AEC36D-9D7D-4C47-B5BC-3A73B208EFA4@gmail.com> <34EA62EA-5133-43A9-AE2A-E8DFD055BEFB@freeswitch.org> Message-ID: <2042CD6C-5F15-4465-9298-E1A2BEECA93D@gmail.com> Thank you. Even the client is broken, we cannot fix that as we don't own the code. But we need that client, is that possible to make FS work around that? On May 5, 2009, at 6:41 PM, Brian West wrote: > It appears to be a broken client. Your client doesn't ack with the > to tag like zoiper does. > > /b > > On May 5, 2009, at 5:10 AM, seven wrote: > >> My problem is why the first one keep sending 404s even got the ACK? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/2f79e2fe/attachment.html From brian at freeswitch.org Tue May 5 03:45:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 5 May 2009 05:45:44 -0500 Subject: [Freeswitch-users] "conf-is-unlocked.wav" missing In-Reply-To: <20090505103525.GA28313@jdc.jasonjgw.net> References: <4A00137A.3070701@gmx.net> <20090505103525.GA28313@jdc.jasonjgw.net> Message-ID: <6FDFB50C-02AE-4429-88C6-1829CB2AF91A@freeswitch.org> The file is absolutely there.. it was just missing the .wav on the end. How hard did you look? :) http://svn.freeswitch.org/svn/sounds/trunk/en/us/callie/48000/conference/conf-is-unlocked.wav I have corrected this in the sounds SVN. /b On May 5, 2009, at 5:35 AM, Jason White wrote: >> conf-is-unlocked.wav Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/43adcd74/attachment.html From Prometheus001 at gmx.net Tue May 5 05:38:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 05 May 2009 14:38:51 +0200 Subject: [Freeswitch-users] "conf-is-unlocked.wav" missing In-Reply-To: <6FDFB50C-02AE-4429-88C6-1829CB2AF91A@freeswitch.org> References: <4A00137A.3070701@gmx.net> <20090505103525.GA28313@jdc.jasonjgw.net> <6FDFB50C-02AE-4429-88C6-1829CB2AF91A@freeswitch.org> Message-ID: <4A00335B.1000700@gmx.net> >How hard did you look? :) I looked at my install directory and in the source files (freeswitch-sounds). No file of this name there. Thanks for the link. Now it works. Best regards Peter Brian West schrieb: > The file is absolutely there.. it was just missing the .wav on the > end. How hard did you look? :) > > http://svn.freeswitch.org/svn/sounds/trunk/en/us/callie/48000/conference/conf-is-unlocked.wav > > I have corrected this in the sounds SVN. > > /b > > > On May 5, 2009, at 5:35 AM, Jason White wrote: > >>> conf-is-unlocked.wav > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Tue May 5 06:52:13 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 05 May 2009 15:52:13 +0200 Subject: [Freeswitch-users] Invite on SIP instead of TLS In-Reply-To: <91A1CA71-4F1C-4B46-9997-3F98159058A0@freeswitch.org> References: <49FF75C6.9060501@gmx.net> <91A1CA71-4F1C-4B46-9997-3F98159058A0@freeswitch.org> Message-ID: <4A00448D.5040805@gmx.net> I updated this. Now TLS invite works. Thank you. Brian West schrieb: > I'm pretty sure this was fixed in 13226 please update. You're using > a new feature it seems. > > /b > > On May 4, 2009, at 6:09 PM, Peter P GMX wrote: > > >> fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 >> > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Tue May 5 07:01:20 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 5 May 2009 09:01:20 -0500 Subject: [Freeswitch-users] Invite on SIP instead of TLS In-Reply-To: <4A00448D.5040805@gmx.net> References: <49FF75C6.9060501@gmx.net> <91A1CA71-4F1C-4B46-9997-3F98159058A0@freeswitch.org> <4A00448D.5040805@gmx.net> Message-ID: <049F5854-AA4E-4884-B1C5-1CFE360EE06B@freeswitch.org> Good to hear! /b On May 5, 2009, at 8:52 AM, Peter P GMX wrote: > I updated this. Now TLS invite works. > > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/7ffbdad8/attachment.html From stevecrozz at gmail.com Tue May 5 07:42:06 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 5 May 2009 07:42:06 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> Message-ID: <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> I know I'm not on svn trunk, but this is a production server and it's just not feasible to update it constantly. I can update it though if you think I need to. I am routing callers to the conference app with javascript like this: session.execute("conference", xyz); Can you tell me more about the problems I could have? The machine running freeswitch has 1024MB memory and I'm not sure about the CPU since its a VPS. --Stephen On Tue, May 5, 2009 at 3:49 AM, Brian West wrote: > First off you're not on SVN trunk secondly Are you executing the conference > app inside your js file? If so then there could be the problem! You have > also forgotten to include anything about Distro, OS, CPU and Memory. > /b > > On May 5, 2009, at 12:12 AM, Stephen Crosby wrote: > > We had our first big issues with our freeswitch system today. During at > least 2 conferences, audio became jittery and there were three occasions > where everyone was dropped from a conference. Even so, conference recording > was not interrupted, and the freeswitch debug log doesn't show anything > unusual. > > Our hardware monitoring software doesn't show any kind of unusual resources > usage, and our web host claims there were no outages during the time when we > were experiencing problems. > > We're currently running revision 12259. > > How should I proceed in diagnosing this issue? > > --Stephen > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/f56eaab8/attachment-0001.html From stevecrozz at gmail.com Tue May 5 07:48:23 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 5 May 2009 07:48:23 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> Message-ID: <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> Forgot to add that my OS is Ubuntu 8.04LTS (hardy heron). --Stephen On Tue, May 5, 2009 at 7:42 AM, Stephen Crosby wrote: > I know I'm not on svn trunk, but this is a production server and it's just > not feasible to update it constantly. I can update it though if you think I > need to. I am routing callers to the conference app with javascript like > this: > session.execute("conference", xyz); > Can you tell me more about the problems I could have? > > The machine running freeswitch has 1024MB memory and I'm not sure about the > CPU since its a VPS. > > --Stephen > > On Tue, May 5, 2009 at 3:49 AM, Brian West wrote: > >> First off you're not on SVN trunk secondly Are you executing the >> conference app inside your js file? If so then there could be the problem! >> You have also forgotten to include anything about Distro, OS, CPU and >> Memory. >> /b >> >> On May 5, 2009, at 12:12 AM, Stephen Crosby wrote: >> >> We had our first big issues with our freeswitch system today. During at >> least 2 conferences, audio became jittery and there were three occasions >> where everyone was dropped from a conference. Even so, conference recording >> was not interrupted, and the freeswitch debug log doesn't show anything >> unusual. >> >> Our hardware monitoring software doesn't show any kind of unusual >> resources usage, and our web host claims there were no outages during the >> time when we were experiencing problems. >> >> We're currently running revision 12259. >> >> How should I proceed in diagnosing this issue? >> >> --Stephen >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/50993178/attachment.html From anthony.minessale at gmail.com Tue May 5 08:25:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 5 May 2009 10:25:26 -0500 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> Message-ID: <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> You should rule out the network problems first, which sound more likely. you can reduce the overuse of JS if you transfer the call to a regular extension with a dynamic regex. session.execute("transfer", "conf-xyz"); then make a regex in your xml dialplan to pick up ^conf-(.*) and execute conference $1 On Tue, May 5, 2009 at 9:48 AM, Stephen Crosby wrote: > Forgot to add that my OS is Ubuntu 8.04LTS (hardy heron). > > --Stephen > > > On Tue, May 5, 2009 at 7:42 AM, Stephen Crosby wrote: > >> I know I'm not on svn trunk, but this is a production server and it's just >> not feasible to update it constantly. I can update it though if you think I >> need to. I am routing callers to the conference app with javascript like >> this: >> session.execute("conference", xyz); >> Can you tell me more about the problems I could have? >> >> The machine running freeswitch has 1024MB memory and I'm not sure about >> the CPU since its a VPS. >> >> --Stephen >> >> On Tue, May 5, 2009 at 3:49 AM, Brian West wrote: >> >>> First off you're not on SVN trunk secondly Are you executing the >>> conference app inside your js file? If so then there could be the problem! >>> You have also forgotten to include anything about Distro, OS, CPU and >>> Memory. >>> /b >>> >>> On May 5, 2009, at 12:12 AM, Stephen Crosby wrote: >>> >>> We had our first big issues with our freeswitch system today. During at >>> least 2 conferences, audio became jittery and there were three occasions >>> where everyone was dropped from a conference. Even so, conference recording >>> was not interrupted, and the freeswitch debug log doesn't show anything >>> unusual. >>> >>> Our hardware monitoring software doesn't show any kind of unusual >>> resources usage, and our web host claims there were no outages during the >>> time when we were experiencing problems. >>> >>> We're currently running revision 12259. >>> >>> How should I proceed in diagnosing this issue? >>> >>> --Stephen >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/7cd36a08/attachment.html From saeedahmad1981 at gmail.com Tue May 5 09:55:17 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 5 May 2009 18:55:17 +0200 Subject: [Freeswitch-users] Inboud Call Queue Message-ID: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. Many Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/c032b05c/attachment.html From msc at freeswitch.org Tue May 5 10:18:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 5 May 2009 10:18:55 -0700 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> Message-ID: <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com> On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed wrote: > Hi All, > > In an inbound call center scenario is it possible that customers calls in > and calls are distributed between online (who are registered on FS and in > idle state) agents. I saw some on going discussion on list where it looks > that currently it?s not possible but I am newbie so maybe I didn?t > understand it well. If it?s possible then please give me a start point that > how can I implement it. > > I would start here: http://wiki.freeswitch.org/wiki/Mod_fifo I strongly recommend that you set up a FreeSWITCH server and play around with it so that you can learn the pros and cons of using the FIFO queues. It would be best if you could set up a few phones and set them as FIFO agents and then have someone help you make test calls so that you can emulate your CC environment. Also, you might want to join us on IRC: #freeswitch on irc.freenode.net - there are several users who've had real world experience with mod_fifo and they might be in a good position to answer your questions real-time. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/6fd9127a/attachment-0001.html From saeedahmad1981 at gmail.com Tue May 5 10:50:47 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 5 May 2009 19:50:47 +0200 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com> Message-ID: <77308CE88F604444863741D590835B10@saeedlaptop> Hi Michael, Thanks for a quick reply. I would definitely create a test environment, but my question is that will it work in required way? I read that in Mod_fifo agent has to call in queue but I need that all incoming calls are automatically distributed between available agents or if all are busy then should go to voicemail. I would join IRC for further assistance. Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 05, 2009 7:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed wrote: Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. I would start here: http://wiki.freeswitch.org/wiki/Mod_fifo I strongly recommend that you set up a FreeSWITCH server and play around with it so that you can learn the pros and cons of using the FIFO queues. It would be best if you could set up a few phones and set them as FIFO agents and then have someone help you make test calls so that you can emulate your CC environment. Also, you might want to join us on IRC: #freeswitch on irc.freenode.net - there are several users who've had real world experience with mod_fifo and they might be in a good position to answer your questions real-time. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/6736a838/attachment.html From Prometheus001 at gmx.net Tue May 5 14:21:20 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 05 May 2009 23:21:20 +0200 Subject: [Freeswitch-users] Invite with TLS when originate Message-ID: <4A00ADD0.9010607@gmx.net> I want to invite another party into a conference with TLS and SRTP enabled. Internal phones are invited by the following dialstring: {originate_timeout=30,sip_secure_media=true,context=default}sofia/default/723321 at sip2.mydomain.de 72332200 Conference'. This enables SRTP but no TLS. Is there any variable I can set in order to enable TLS? set internal_auth_calls=true is meant only for configuration, hein? Also context=default doesn't succeed in this case. The call is passed to the public context. Best regards Peter From brian at freeswitch.org Tue May 5 14:27:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 5 May 2009 16:27:02 -0500 Subject: [Freeswitch-users] Invite with TLS when originate In-Reply-To: <4A00ADD0.9010607@gmx.net> References: <4A00ADD0.9010607@gmx.net> Message-ID: <1CDFB25B-D719-45A2-86E3-39CAA7CC8662@freeswitch.org> now append transport=tls > {originate_timeout=30,sip_secure_media=true,context=default}sofia/default/723321 at sip2.mydomain.de > ;transport=tls Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/e635fda4/attachment.html From Prometheus001 at gmx.net Tue May 5 15:06:13 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 06 May 2009 00:06:13 +0200 Subject: [Freeswitch-users] Invite with TLS when originate In-Reply-To: <1CDFB25B-D719-45A2-86E3-39CAA7CC8662@freeswitch.org> References: <4A00ADD0.9010607@gmx.net> <1CDFB25B-D719-45A2-86E3-39CAA7CC8662@freeswitch.org> Message-ID: <4A00B855.8000805@gmx.net> When I append transport=tls I recieve the following and the call is not initiated: 2009-05-06 00:01:37 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/723321 at sip2.mydomain.de;transport=tls SOFIA INIT 2009-05-06 00:01:37 [DEBUG] sofia_glue.c:1972 sofia_glue_build_crypto() Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:AfDrMfXhTLFqPVOvzwTNV+9Wa8WYdh/TiSlZ90f0] 2009-05-06 00:01:38 [DEBUG] sofia_glue.c:583 sofia_glue_ext_address_lookup() STUN Success [217.xxx.xxx.190]:[12300] 2009-05-06 00:01:38 [DEBUG] sofia_glue.c:587 sofia_glue_ext_address_lookup() STUN Not Required ip and port match. [217.xxx.xxx.190]:[12300] 2009-05-06 00:01:38 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) State Change CS_INIT -> CS_ROUTING 2009-05-06 00:01:38 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/723321 at sip2.mydomain.de;transport=tls [BREAK] 2009-05-06 00:01:38 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) State INIT going to sleep 2009-05-06 00:01:38 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) Running State Change CS_ROUTING 2009-05-06 00:01:38 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) State ROUTING 2009-05-06 00:01:38 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/723321 at sip2.mydomain.de;transport=tls SOFIA ROUTING 2009-05-06 00:01:38 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-05-06 00:01:38 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/723321 at sip2.mydomain.de;transport=tls [BREAK] 2009-05-06 00:01:38 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) State ROUTING going to sleep 2009-05-06 00:01:38 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) Running State Change CS_CONSUME_MEDIA 2009-05-06 00:01:38 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/723321 at sip2.mydomain.de;transport=tls) State CONSUME_MEDIA 2009-05-06 00:01:38 [DEBUG] sofia.c:2911 sofia_handle_sip_i_state() Channel sofia/internal/723321 at sip2.mydomain.de;transport=tls entering state [calling][0] 2009-05-06 00:01:38 [DEBUG] sofia.c:4241 sofia_handle_sip_i_invite() IP 217.xxx.xxx.190 Rejected by acl "domains". Falling back to Digest auth. 2009-05-06 00:01:38 [ERR] sofia_reg.c:1489 sofia_reg_handle_sip_r_challenge() No Matching gateway found 2009-05-06 00:01:38 [NOTICE] sofia_reg.c:1508 sofia_reg_handle_sip_r_challenge() Hangup sofia/internal/723321 at sip2.mydomain.de;transport=tls [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] 2009-05-06 00:01:38 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal sofia/internal/723321 at sip2.mydomain.de;transport=tls [KILL] 2009-05-06 00:01:38 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/723321 at sip2.mydomain.de;transport=tls [BREAK] 2009-05-06 00:01:38 [DEBUG] switch_ivr_originate.c:2094 switch_ivr_originate() Originate Resulted in Error Cause: 96 [MANDATORY_IE_MISSING] 2009-05-06 00:01:38 [ERR] mod_conference.c:4326 conference_outcall() Cannot create outgoing channel, cause: MANDATORY_IE_MISSING Brian West schrieb: > now append transport=tls > >> {originate_timeout=30,sip_secure_media=true,context=default}sofia/default/723321 at sip2.mydomain.de >> ;transport=tls > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue May 5 15:10:47 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 5 May 2009 17:10:47 -0500 Subject: [Freeswitch-users] Invite with TLS when originate In-Reply-To: <4A00B855.8000805@gmx.net> References: <4A00ADD0.9010607@gmx.net> <1CDFB25B-D719-45A2-86E3-39CAA7CC8662@freeswitch.org> <4A00B855.8000805@gmx.net> Message-ID: <7E5A5F51-E851-40C8-ADFC-FB0AD3E59054@freeswitch.org> The far end challenged you and it looks like you couldn't answer said challenge. /b On May 5, 2009, at 5:06 PM, Peter P GMX wrote: > Cannot create outgoing channel, cause: MANDATORY_IE_MISSING Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/ee341fa7/attachment.html From Prometheus001 at gmx.net Tue May 5 15:48:10 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 06 May 2009 00:48:10 +0200 Subject: [Freeswitch-users] Invite with TLS when originate In-Reply-To: <7E5A5F51-E851-40C8-ADFC-FB0AD3E59054@freeswitch.org> References: <4A00ADD0.9010607@gmx.net> <1CDFB25B-D719-45A2-86E3-39CAA7CC8662@freeswitch.org> <4A00B855.8000805@gmx.net> <7E5A5F51-E851-40C8-ADFC-FB0AD3E59054@freeswitch.org> Message-ID: <4A00C22A.5000309@gmx.net> The far end is a Snom phone which I can dial the normal way (Snom -> FS -> Snom) via TLS. So I have no clue what to do now. Any hint? Best regards Peter Brian West schrieb: > The far end challenged you and it looks like you couldn't answer said > challenge. > > /b > > On May 5, 2009, at 5:06 PM, Peter P GMX wrote: > >> Cannot create outgoing channel, cause: MANDATORY_IE_MISSING > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Tue May 5 17:48:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 5 May 2009 17:48:21 -0700 Subject: [Freeswitch-users] ClueCon 2009 Blog, Moises Silva Speaking Message-ID: <87f2f3b90905051748s788914c7x6fdd557be342f64a@mail.gmail.com> FYI, I just wanted to let the community know that we maintain a blogon the ClueCon website . Please check it out. The latest entry mentions Moises Silva's recent blog entry about his speaking at ClueCon this year. Please check the ClueCon blog periodically as we will be adding new information about speakers, sponsors, and other good stuff. -Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090505/6cd86c66/attachment.html From dujinfang at gmail.com Tue May 5 19:17:12 2009 From: dujinfang at gmail.com (seven) Date: Wed, 6 May 2009 10:17:12 +0800 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <77308CE88F604444863741D590835B10@saeedlaptop> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com> <77308CE88F604444863741D590835B10@saeedlaptop> Message-ID: <63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com> On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: > Hi Michael, > > Thanks for a quick reply. > > I would definitely create a test environment, but my question is > that will it work in required way? > > I read that in Mod_fifo agent has to call in queue but I need that > all incoming calls are automatically distributed between available > agents or if all are busy then should go to voicemail. > I'm working on a call center like queue scenario right now, I'm pretty sure it call automatically distributed to available agents, but the customer will stay in the queue if all agents are busy by default. You can bind a key to the channel and play a message repeatedly to guide the customer to voicemail by press a key. Also maybe you need this patch to make the fifo works as desired. http://jira.freeswitch.org/browse/MODAPP-272 > I would join IRC for further assistance. > > Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, May 05, 2009 7:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Inboud Call Queue > > > On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed > wrote: > Hi All, > > In an inbound call center scenario is it possible that customers > calls in and calls are distributed between online (who are > registered on FS and in idle state) agents. I saw some on going > discussion on list where it looks that currently it?s not possible > but I am newbie so maybe I didn?t understand it well. If it?s > possible then please give me a start point that how can I implement > it. > > I would start here: > http://wiki.freeswitch.org/wiki/Mod_fifo > > I strongly recommend that you set up a FreeSWITCH server and play > around with it so that you can learn the pros and cons of using the > FIFO queues. It would be best if you could set up a few phones and > set them as FIFO agents and then have someone help you make test > calls so that you can emulate your CC environment. > > Also, you might want to join us on IRC: #freeswitch on > irc.freenode.net - there are several users who've had real world > experience with mod_fifo and they might be in a good position to > answer your questions real-time. > > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/f970b419/attachment.html From exyeechen at hotmail.com Tue May 5 22:22:08 2009 From: exyeechen at hotmail.com (chenexyee) Date: Wed, 6 May 2009 13:22:08 +0800 Subject: [Freeswitch-users] Busy tone and text message configuration Message-ID: Hi all I'm not familiar with the configuration for freeswitch. Anyone who could help me sovle the following two problem: 1. user A is in conversation with user B, and at this time, a incoming call from user C comes to A, in this case, I want freeswitch to play busytone to C, how to configure? 2. I'd like freeswitch to relay text message(use sip),as below scenario: entity1--------------------------freeswitch---------------------------entity2 | message(text/plain) ------> | message(text/plain) ------> | | <-------------200 OK | <-------------200 OK | | | | To implement above case, does need any special configure for freeswitch? Appreciate your help in advance. _________________________________________________________________ ????Live????Windows Live????, ???????? http://events.livetome.cn/2009/knowlive -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/7bf28b41/attachment.html From mike at jerris.com Tue May 5 23:09:25 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 6 May 2009 02:09:25 -0400 Subject: [Freeswitch-users] "conf-is-unlocked.wav" missing In-Reply-To: <4A00335B.1000700@gmx.net> References: <4A00137A.3070701@gmx.net> <20090505103525.GA28313@jdc.jasonjgw.net> <6FDFB50C-02AE-4429-88C6-1829CB2AF91A@freeswitch.org> <4A00335B.1000700@gmx.net> Message-ID: <79484460-D811-4C63-B5E1-1F7A048B7A94@jerris.com> the 1.0.9 sounds were rolled tonight and they contain these fixes. Mike On May 5, 2009, at 8:38 AM, Peter P GMX wrote: > I looked at my install directory and in the source files > (freeswitch-sounds). No file of this name there. > > Thanks for the link. Now it works. > > Best regards > Peter > > Brian West schrieb: >> The file is absolutely there.. it was just missing the .wav on the >> end. How hard did you look? :) >> >> http://svn.freeswitch.org/svn/sounds/trunk/en/us/callie/48000/conference/conf-is-unlocked.wav >> >> I have corrected this in the sounds SVN. >> >> /b >> >> >> On May 5, 2009, at 5:35 AM, Jason White wrote: >> >>>> conf-is-unlocked.wav >> >> ---------------------------------- Check out the Barracuda Spam & Virus Firewall - offering the fastest virus & malware protection in the industry: www.barracudanetworks.com/spam From jonas.gauffin at gmail.com Tue May 5 23:36:05 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 6 May 2009 08:36:05 +0200 Subject: [Freeswitch-users] Profile reloading Message-ID: Hello, I have to static IP:s on my server. FS has been bound to one of them. Yesterday evening I got these log messages: 2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler() IP change detected [85.89.XX.XX9]->[85.89.XX.XX8] []->[] 2009-05-05 19:17:37 [NOTICE] sofia_glue.c:3303 sofia_glue_restart_all_profiles() Reload XML [Success] And since FS changed IP to the other one, none of the phones could register to FS. The problem is that both IP addresses are the same ones that the server always have had. And XX9 is the one that I've bound freeswitch too. Regards, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/aff26ef4/attachment.html From saeedahmad1981 at gmail.com Wed May 6 03:45:42 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 6 May 2009 12:45:42 +0200 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop><87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com><77308CE88F604444863741D590835B10@saeedlaptop> <63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com> Message-ID: Hi Seven, I am exactly looking for this functionality. Please let me know when you are finished with new queue manager app. I'll try it in my call center. Regarding Patch: is it already part of SVN trunk? If not then could you help me how to install it, I have no programming background. Many Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of seven Sent: Wednesday, May 06, 2009 4:17 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: Hi Michael, Thanks for a quick reply. I would definitely create a test environment, but my question is that will it work in required way? I read that in Mod_fifo agent has to call in queue but I need that all incoming calls are automatically distributed between available agents or if all are busy then should go to voicemail. I'm working on a call center like queue scenario right now, I'm pretty sure it call automatically distributed to available agents, but the customer will stay in the queue if all agents are busy by default. You can bind a key to the channel and play a message repeatedly to guide the customer to voicemail by press a key. Also maybe you need this patch to make the fifo works as desired. http://jira.freeswitch.org/browse/MODAPP-272 I would join IRC for further assistance. Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 05, 2009 7:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed wrote: Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. I would start here: http://wiki.freeswitch.org/wiki/Mod_fifo I strongly recommend that you set up a FreeSWITCH server and play around with it so that you can learn the pros and cons of using the FIFO queues. It would be best if you could set up a few phones and set them as FIFO agents and then have someone help you make test calls so that you can emulate your CC environment. Also, you might want to join us on IRC: #freeswitch on irc.freenode.net - there are several users who've had real world experience with mod_fifo and they might be in a good position to answer your questions real-time. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/36530755/attachment-0001.html From brian at freeswitch.org Wed May 6 05:01:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 6 May 2009 07:01:49 -0500 Subject: [Freeswitch-users] Profile reloading In-Reply-To: References: Message-ID: <9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org> add to conf/ autoload_configs/sofia.conf.xml Did you happen to bind the IP while FS was running? /b On May 6, 2009, at 1:36 AM, Jonas Gauffin wrote: > Hello, > > I have to static IP:s on my server. FS has been bound to one of them. > Yesterday evening I got these log messages: > > 2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler() > IP change detected [85.89.XX.XX9]->[85.89.XX.XX8] []->[] > 2009-05-05 19:17:37 [NOTICE] sofia_glue.c:3303 > sofia_glue_restart_all_profiles() Reload XML [Success] > > And since FS changed IP to the other one, none of the phones could > register to FS. > > The problem is that both IP addresses are the same ones that the > server always have had. And XX9 is the one that I've bound > freeswitch too. > > Regards, > Jonas Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/a691ea2b/attachment.html From dujinfang at gmail.com Wed May 6 05:53:40 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 6 May 2009 20:53:40 +0800 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop><87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com><77308CE88F604444863741D590835B10@saeedlaptop> <63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com> Message-ID: <18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com> The patch haven't been merged into trunk. It should be as easy as execute the following command in the FS source code root dir: patch < /tmp/the_patch_file_name.diff I will post an example on the wiki when I finished, hope be soon. On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: > Hi Seven, > > I am exactly looking for this functionality. > > Please let me know when you are finished with new queue manager app. > I?ll try it in my call center. > > Regarding Patch: is it already part of SVN trunk? If not then could > you help me how to install it, I have no programming background. > > Many Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of seven > Sent: Wednesday, May 06, 2009 4:17 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Inboud Call Queue > > > On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: > > > Hi Michael, > > Thanks for a quick reply. > > I would definitely create a test environment, but my question is > that will it work in required way? > > I read that in Mod_fifo agent has to call in queue but I need that > all incoming calls are automatically distributed between available > agents or if all are busy then should go to voicemail. > I'm working on a call center like queue scenario right now, I'm > pretty sure it call automatically distributed to available agents, > but the customer will stay in the queue if all agents are busy by > default. You can bind a key to the channel and play a message > repeatedly to guide the customer to voicemail by press a key. > > Also maybe you need this patch to make the fifo works as desired. > > http://jira.freeswitch.org/browse/MODAPP-272 > > > I would join IRC for further assistance. > > Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, May 05, 2009 7:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Inboud Call Queue > > > On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed > wrote: > Hi All, > > In an inbound call center scenario is it possible that customers > calls in and calls are distributed between online (who are > registered on FS and in idle state) agents. I saw some on going > discussion on list where it looks that currently it?s not possible > but I am newbie so maybe I didn?t understand it well. If it?s > possible then please give me a start point that how can I implement > it. > > I would start here: > http://wiki.freeswitch.org/wiki/Mod_fifo > > I strongly recommend that you set up a FreeSWITCH server and play > around with it so that you can learn the pros and cons of using the > FIFO queues. It would be best if you could set up a few phones and > set them as FIFO agents and then have someone help you make test > calls so that you can emulate your CC environment. > > Also, you might want to join us on IRC: #freeswitch on > irc.freenode.net - there are several users who've had real world > experience with mod_fifo and they might be in a good position to > answer your questions real-time. > > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/bec37de7/attachment-0001.html From jonas.gauffin at gmail.com Wed May 6 05:57:01 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 6 May 2009 14:57:01 +0200 Subject: [Freeswitch-users] Profile reloading In-Reply-To: <9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org> References: <9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org> Message-ID: No, have never changed the IPs since the server was installed. And have not changed it in FS either. Ok. Will add the parameter. Thanks. On Wed, May 6, 2009 at 2:01 PM, Brian West wrote: > add > to conf/autoload_configs/sofia.conf.xml > Did you happen to bind the IP while FS was running? > > /b > > On May 6, 2009, at 1:36 AM, Jonas Gauffin wrote: > > Hello, > I have to static IP:s on my server. FS has been bound to one of them. > Yesterday evening I got these log messages: > > 2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler() IP > change detected [85.89.XX.XX9]->[85.89.XX.XX8] []->[] > 2009-05-05 19:17:37 [NOTICE] sofia_glue.c:3303 > sofia_glue_restart_all_profiles() Reload XML [Success] > > And since FS changed IP to the other one, none of the phones could register > to FS. > > The problem is that both IP addresses are the same ones that the server > always have had. And XX9 is the one that I've bound freeswitch too. > > Regards, > Jonas > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/9fea8d7f/attachment.html From brian at freeswitch.org Wed May 6 06:00:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 6 May 2009 08:00:39 -0500 Subject: [Freeswitch-users] Profile reloading In-Reply-To: References: <9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org> Message-ID: <5DD52D3F-DA80-475D-BC13-7071EA780E33@freeswitch.org> The IP guessing code changed its guess then... /b On May 6, 2009, at 7:57 AM, Jonas Gauffin wrote: > No, have never changed the IPs since the server was installed. And > have not changed it in FS either. > > Ok. Will add the parameter. Thanks. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/f777ec4a/attachment.html From intralanman at freeswitch.org Wed May 6 06:47:42 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 06 May 2009 09:47:42 -0400 Subject: [Freeswitch-users] Busy tone and text message configuration In-Reply-To: References: Message-ID: <4A0194FE.9030609@freeswitch.org> chenexyee wrote: > > 1. user A is in conversation with user B, and at this time, a incoming > call from user C comes to A, in this case, I want freeswitch to play > busytone to C, how to configure? you could use the limit app (mod_limit) to limit A's number of calls to 1, then play the busy sound with tone_stream or return a 486 to the caller in the failover extension > > 2. I'd like freeswitch to relay text message(use sip),as below scenario: DISCLAIMER: i'm speaking purely theory here as i've never tried to do this. but you could probably write something to listen via esl for the incoming event, then in your program, send that event to the user you want it relayed to. From anthony.minessale at gmail.com Wed May 6 06:57:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 6 May 2009 08:57:07 -0500 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com> <77308CE88F604444863741D590835B10@saeedlaptop> <63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com> <18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com> Message-ID: <191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com> I worked on the patch and added it to trunk rev 13240 On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote: > The patch haven't been merged into trunk. It should be as easy as execute > the following command in the FS source code root dir: > patch < /tmp/the_patch_file_name.diff > > I will post an example on the wiki when I finished, hope be soon. > > On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: > > Hi Seven, > > I am exactly looking for this functionality. > > Please let me know when you are finished with new queue manager app. I?ll > try it in my call center. > > Regarding Patch: is it already part of SVN trunk? If not then could you > help me how to install it, I have no programming background. > > Many Thanks. > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *seven > *Sent:* Wednesday, May 06, 2009 4:17 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Inboud Call Queue > > > On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: > > > Hi Michael, > > Thanks for a quick reply. > > I would definitely create a test environment, but my question is that will > it work in required way? > > I read that in Mod_fifo agent has to call in queue but I need that all > incoming calls are automatically distributed between available agents or if > all are busy then should go to voicemail. > I'm working on a call center like queue scenario right now, I'm pretty sure > it call automatically distributed to available agents, but the customer will > stay in the queue if all agents are busy by default. You can bind a key to > the channel and play a message repeatedly to guide the customer to voicemail > by press a key. > > Also maybe you need this patch to make the fifo works as desired. > > http://jira.freeswitch.org/browse/MODAPP-272 > > > I would join IRC for further assistance. > > Thanks. > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Michael Collins > *Sent:* Tuesday, May 05, 2009 7:19 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Inboud Call Queue > > > On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed > wrote: > > Hi All, > > In an inbound call center scenario is it possible that customers calls in > and calls are distributed between online (who are registered on FS and in > idle state) agents. I saw some on going discussion on list where it looks > that currently it?s not possible but I am newbie so maybe I didn?t > understand it well. If it?s possible then please give me a start point that > how can I implement it. > I would start here: > http://wiki.freeswitch.org/wiki/Mod_fifo > > I strongly recommend that you set up a FreeSWITCH server and play around > with it so that you can learn the pros and cons of using the FIFO queues. It > would be best if you could set up a few phones and set them as FIFO agents > and then have someone help you make test calls so that you can emulate your > CC environment. > > Also, you might want to join us on IRC: #freeswitch on irc.freenode.net - > there are several users who've had real world experience with mod_fifo and > they might be in a good position to answer your questions real-time. > > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/3495a1aa/attachment-0001.html From msc at freeswitch.org Wed May 6 07:06:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 May 2009 07:06:03 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre7 Now Available Message-ID: <87f2f3b90905060706j7c8d4d2dh5bd620db65ab52a@mail.gmail.com> FYI, Please update your installations as soon as possible. More information on this update is available here . Thanks for all of your feedback - please keep it coming and join us on IRC if you have any questions about the newest version. -Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/c4818bf8/attachment.html From dujinfang at gmail.com Wed May 6 07:11:30 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 6 May 2009 22:11:30 +0800 Subject: [Freeswitch-users] Busy tone and text message configuration In-Reply-To: <4A0194FE.9030609@freeswitch.org> References: <4A0194FE.9030609@freeswitch.org> Message-ID: <6A31DE88-C357-4167-924C-346CBCAAEAC9@gmail.com> On May 6, 2009, at 9:47 PM, Raymond Chandler wrote: > chenexyee wrote: >> >> 1. user A is in conversation with user B, and at this time, a >> incoming >> call from user C comes to A, in this case, I want freeswitch to play >> busytone to C, how to configure? > you could use the limit app (mod_limit) to limit A's number of calls > to > 1, then play the busy sound with tone_stream or return a 486 to the > caller in the failover extension I'm also finding a way to limit only one call to user A, however I think mod_limit is used to limit outbound calls only, how can it possible to limit incoming calls for a user? > >> >> 2. I'd like freeswitch to relay text message(use sip),as below >> scenario: > DISCLAIMER: i'm speaking purely theory here as i've never tried to > do this. > but you could probably write something to listen via esl for the > incoming event, then in your program, send that event to the user you > want it relayed to. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed May 6 07:15:13 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 6 May 2009 09:15:13 -0500 Subject: [Freeswitch-users] Busy tone and text message configuration In-Reply-To: <6A31DE88-C357-4167-924C-346CBCAAEAC9@gmail.com> References: <4A0194FE.9030609@freeswitch.org> <6A31DE88-C357-4167-924C-346CBCAAEAC9@gmail.com> Message-ID: The exact same way you use it for outbound... just use limit before you call the user in your dial plan. An inbound call to a user is nothing more than an outbound call from FreeSWITCH to the user. /b On May 6, 2009, at 9:11 AM, dujinfang wrote: > I'm also finding a way to limit only one call to user A, however I > think mod_limit is used to limit outbound calls only, how can it > possible to limit incoming calls for a user? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/34330d35/attachment.html From saeedahmad1981 at gmail.com Wed May 6 07:15:54 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 6 May 2009 16:15:54 +0200 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop><87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com><77308CE88F604444863741D590835B10@saeedlaptop><63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com><18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com> <191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com> Message-ID: <62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop> Thanks Guys _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, May 06, 2009 3:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue I worked on the patch and added it to trunk rev 13240 On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote: The patch haven't been merged into trunk. It should be as easy as execute the following command in the FS source code root dir: patch < /tmp/the_patch_file_name.diff I will post an example on the wiki when I finished, hope be soon. On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: Hi Seven, I am exactly looking for this functionality. Please let me know when you are finished with new queue manager app. I'll try it in my call center. Regarding Patch: is it already part of SVN trunk? If not then could you help me how to install it, I have no programming background. Many Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of seven Sent: Wednesday, May 06, 2009 4:17 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: Hi Michael, Thanks for a quick reply. I would definitely create a test environment, but my question is that will it work in required way? I read that in Mod_fifo agent has to call in queue but I need that all incoming calls are automatically distributed between available agents or if all are busy then should go to voicemail. I'm working on a call center like queue scenario right now, I'm pretty sure it call automatically distributed to available agents, but the customer will stay in the queue if all agents are busy by default. You can bind a key to the channel and play a message repeatedly to guide the customer to voicemail by press a key. Also maybe you need this patch to make the fifo works as desired. http://jira.freeswitch.org/browse/MODAPP-272 I would join IRC for further assistance. Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 05, 2009 7:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed wrote: Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. I would start here: http://wiki.freeswitch.org/wiki/Mod_fifo I strongly recommend that you set up a FreeSWITCH server and play around with it so that you can learn the pros and cons of using the FIFO queues. It would be best if you could set up a few phones and set them as FIFO agents and then have someone help you make test calls so that you can emulate your CC environment. Also, you might want to join us on IRC: #freeswitch on irc.freenode.net - there are several users who've had real world experience with mod_fifo and they might be in a good position to answer your questions real-time. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/19b7511f/attachment-0001.html From dujinfang at gmail.com Wed May 6 07:23:33 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 6 May 2009 22:23:33 +0800 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop> <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com> <77308CE88F604444863741D590835B10@saeedlaptop> <63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com> <18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com> <191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com> Message-ID: <4F034D4E-A817-4D90-8AFB-0DA92A45E706@gmail.com> Thanks, so quick. Actually I had submitted another version of patch which added a channel var fifo_caller_exit_to_orbit which make the caller possible to exit to the orbit_exten other than hangup the caller when the caller enter the fifo_caller_exit_key. I use this to guide the caller to another ivr or voice mail for non- patient callers. If you think that useful, I can add another patch to jira. Apparently if the fifo can bind to more keys like ivr does will be better, in that way it can give callers more options and we can play announcement by fifo_chime_list. - if (cd.do_orbit && cd.orbit_exten) { + if ((switch_true(switch_channel_get_variable(channel, "fifo_caller_exit_to_orbit")) || cd.do_orbit) && cd.orbit_exten) { On May 6, 2009, at 9:57 PM, Anthony Minessale wrote: > I worked on the patch and added it to trunk rev 13240 > > > On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote: > The patch haven't been merged into trunk. It should be as easy as > execute the following command in the FS source code root dir: > > patch < /tmp/the_patch_file_name.diff > > I will post an example on the wiki when I finished, hope be soon. > > On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: >> Hi Seven, >> >> I am exactly looking for this functionality. >> >> Please let me know when you are finished with new queue manager >> app. I?ll try it in my call center. >> >> Regarding Patch: is it already part of SVN trunk? If not then could >> you help me how to install it, I have no programming background. >> >> Many Thanks. >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of seven >> Sent: Wednesday, May 06, 2009 4:17 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Inboud Call Queue >> >> >> On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: >> >> >> Hi Michael, >> >> Thanks for a quick reply. >> >> I would definitely create a test environment, but my question is >> that will it work in required way? >> >> I read that in Mod_fifo agent has to call in queue but I need that >> all incoming calls are automatically distributed between available >> agents or if all are busy then should go to voicemail. >> I'm working on a call center like queue scenario right now, I'm >> pretty sure it call automatically distributed to available agents, >> but the customer will stay in the queue if all agents are busy by >> default. You can bind a key to the channel and play a message >> repeatedly to guide the customer to voicemail by press a key. >> >> Also maybe you need this patch to make the fifo works as desired. >> >> http://jira.freeswitch.org/browse/MODAPP-272 >> >> >> I would join IRC for further assistance. >> >> Thanks. >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Michael Collins >> Sent: Tuesday, May 05, 2009 7:19 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Inboud Call Queue >> >> >> On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed >> wrote: >> Hi All, >> >> In an inbound call center scenario is it possible that customers >> calls in and calls are distributed between online (who are >> registered on FS and in idle state) agents. I saw some on going >> discussion on list where it looks that currently it?s not possible >> but I am newbie so maybe I didn?t understand it well. If it?s >> possible then please give me a start point that how can I implement >> it. >> >> I would start here: >> http://wiki.freeswitch.org/wiki/Mod_fifo >> >> I strongly recommend that you set up a FreeSWITCH server and play >> around with it so that you can learn the pros and cons of using the >> FIFO queues. It would be best if you could set up a few phones and >> set them as FIFO agents and then have someone help you make test >> calls so that you can emulate your CC environment. >> >> Also, you might want to join us on IRC: #freeswitch on >> irc.freenode.net - there are several users who've had real world >> experience with mod_fifo and they might be in a good position to >> answer your questions real-time. >> >> -MC >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/9ca26905/attachment.html From moizchinoy at gmail.com Wed May 6 07:58:35 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 6 May 2009 19:58:35 +0500 Subject: [Freeswitch-users] Sphinx 4 Integration... Message-ID: <29b888f80905060758i489ca8e3ned44e405ab7081e3@mail.gmail.com> Hi All, Has anyone tried Zanzibar and Cairo. They have implemented MRCP 2.0 and integrated Sphinx 4. Since MRCP 2.0 supports SIP for communication, it can easily be integrated with FS! Although it is implemented in Java, VoiceXML is also supported! -- Regards, Moiz Chinoy. From brian at freeswitch.org Wed May 6 08:11:46 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 6 May 2009 10:11:46 -0500 Subject: [Freeswitch-users] Sphinx 4 Integration... In-Reply-To: <29b888f80905060758i489ca8e3ned44e405ab7081e3@mail.gmail.com> References: <29b888f80905060758i489ca8e3ned44e405ab7081e3@mail.gmail.com> Message-ID: <34611AE0-3F0C-4198-9872-AE7A28BA6E55@freeswitch.org> You're a bit mistaken on the MRCP 2.0 supporting SIP.. it uses SIP for signaling and RTP for media transport. ... that however doesn't mean you can just call it via SIP and it work. They put a little bit more goop on top of that! /b On May 6, 2009, at 9:58 AM, Moiz Chinoy wrote: > Hi All, > > Has anyone tried Zanzibar and Cairo. They have implemented MRCP 2.0 > and integrated Sphinx 4. Since MRCP 2.0 supports SIP for > communication, it can easily be integrated with FS! > > Although it is implemented in Java, VoiceXML is also supported! > -- > Regards, > Moiz Chinoy. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/7d3017f4/attachment-0001.html From moizchinoy at gmail.com Wed May 6 08:38:29 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 6 May 2009 20:38:29 +0500 Subject: [Freeswitch-users] Sphinx 4 Integration... In-Reply-To: <34611AE0-3F0C-4198-9872-AE7A28BA6E55@freeswitch.org> References: <29b888f80905060758i489ca8e3ned44e405ab7081e3@mail.gmail.com> <34611AE0-3F0C-4198-9872-AE7A28BA6E55@freeswitch.org> Message-ID: <29b888f80905060838v23654b4do91eacf6839d49713@mail.gmail.com> Yes you are right, it uses SIP for signaling and RTP for media transport. But does it make any difference because in its documentation they have stated that they support Asterisk and any IP PBX that supports SIP and RTP. I haven't tried it with FS yet but it worked with Xlite using SIP. For some reason it failed with SjPhone giving SDP error. On Wed, May 6, 2009 at 8:11 PM, Brian West wrote: > You're a bit mistaken on the MRCP 2.0 supporting SIP.. it uses SIP for > signaling and RTP for media transport. ... that however doesn't mean you can > just call it via SIP and it work. ?They put a little bit more goop on top of > that! > /b > On May 6, 2009, at 9:58 AM, Moiz Chinoy wrote: > > Hi All, > > Has anyone tried Zanzibar and Cairo. They have implemented MRCP 2.0 > and integrated Sphinx 4. Since MRCP 2.0 supports SIP for > communication, it can easily be integrated with FS! > > Although it is implemented in Java, VoiceXML is also supported! > -- > Regards, > Moiz Chinoy. > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Moiz Chinoy. From msc at freeswitch.org Wed May 6 09:36:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 May 2009 09:36:34 -0700 Subject: [Freeswitch-users] Interesting Blog About HD Telephony Message-ID: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> FYI, Here's a nice story for you all to check out. Please check it out and pass it on. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/0b40af8d/attachment.html From gmaruzz at celliax.org Wed May 6 10:00:57 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 6 May 2009 19:00:57 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony In-Reply-To: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> Message-ID: <7b197bef0905061000v56269cceh73c8b7519662275e@mail.gmail.com> Ciao Michael, if you like, you can add that using mod_skypiax you have native hd skype->FS and FS->Skype (no hardware needed) :-) On Wed, May 6, 2009 at 6:36 PM, Michael Collins wrote: > FYI, > > Here's a nice story for you all to check out. Please check it out and pass > it on. > > -Michael > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > From msc at freeswitch.org Wed May 6 11:23:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 May 2009 11:23:46 -0700 Subject: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony In-Reply-To: <7b197bef0905061000v56269cceh73c8b7519662275e@mail.gmail.com> References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> <7b197bef0905061000v56269cceh73c8b7519662275e@mail.gmail.com> Message-ID: <87f2f3b90905061123h19f32216p235ff71e5e0db44@mail.gmail.com> On Wed, May 6, 2009 at 10:00 AM, Giovanni Maruzzelli wrote: > Ciao Michael, > if you like, you can add that using mod_skypiax you have native hd > skype->FS and FS->Skype (no hardware needed) :-) > > FYI, I made a comment on Dave's blog extolling the virtues of FS and I mentioned Skype support. I didn't specifically mention mod_skypiax but I didn't specifically mention any mods. Anyway, let's see what kind interest we see brewing in HD Voice. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/a3b287b0/attachment.html From gmaruzz at celliax.org Wed May 6 11:35:12 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 6 May 2009 20:35:12 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony In-Reply-To: <87f2f3b90905061123h19f32216p235ff71e5e0db44@mail.gmail.com> References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> <7b197bef0905061000v56269cceh73c8b7519662275e@mail.gmail.com> <87f2f3b90905061123h19f32216p235ff71e5e0db44@mail.gmail.com> Message-ID: <7b197bef0905061135s62ea2cb4hfd0fb2fc8bacace7@mail.gmail.com> On Wed, May 6, 2009 at 8:23 PM, Michael Collins wrote: > FYI, > I made a comment on Dave's blog extolling the virtues of FS and I mentioned > Skype support. I didn't specifically mention mod_skypiax but I didn't > specifically mention any mods. > I was suggesting to put mod_skypiax in the http://www.freeswitch.org/node/182 page, for ourselves BTW: Very nice comment, it sure will attract attention! -gm From asobihoudai at yahoo.com Wed May 6 11:38:00 2009 From: asobihoudai at yahoo.com (Paul) Date: Wed, 6 May 2009 11:38:00 -0700 (PDT) Subject: [Freeswitch-users] Interesting Blog About HD Telephony In-Reply-To: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> Message-ID: <506966.3998.qm@web111309.mail.gq1.yahoo.com> Are the most currently ratified HD Voice codecs G.722 and G.722.1? I haven't heard very much about HD Voice at all until you just brought it up. ________________________________ From: Michael Collins To: "freeswitch-users at lists.freeswitch.org" ; freeswitch-dev at lists.freeswitch.org Sent: Wednesday, May 6, 2009 12:36:34 PM Subject: [Freeswitch-users] Interesting Blog About HD Telephony FYI, Here's a nice story for you all to check out. Please check it out and pass it on. -Michael From brian at freeswitch.org Wed May 6 11:40:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 6 May 2009 13:40:37 -0500 Subject: [Freeswitch-users] Interesting Blog About HD Telephony In-Reply-To: <506966.3998.qm@web111309.mail.gq1.yahoo.com> References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> <506966.3998.qm@web111309.mail.gq1.yahoo.com> Message-ID: <24A0E132-CAEA-4182-8A29-4C38DF78D4AD@freeswitch.org> Well its not so much which codecs but that the codecs can do 16k... Currently FreesWITCH supports DVI4 at 16k, Speex at 16k, G722 and G722.1 at 16k But as a bonus: We can also do Speex at 32k, G722.1C at 32k, Celt at 32k and 48k /b On May 6, 2009, at 1:38 PM, Paul wrote: > > Are the most currently ratified HD Voice codecs G.722 and G.722.1? I > haven't heard very much about HD Voice at all until you just brought > it up. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/bbbbd84b/attachment.html From austad at signal15.com Wed May 6 11:44:37 2009 From: austad at signal15.com (Jay Austad) Date: Wed, 6 May 2009 13:44:37 -0500 Subject: [Freeswitch-users] DTMF recognition flaky Message-ID: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com> Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? -- jay austad | 612.423.1433 | austad at signal15.com From brian at freeswitch.org Wed May 6 11:46:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 6 May 2009 13:46:55 -0500 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com> References: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com> Message-ID: Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote: > Using the default installation, I've noticed that when I (or someone > else) calls in on my SIP trunk and keys in an extension, not all of > the numbers are recognized unless they hold the key down for at least > 1/2 second to a second. > > Is there a way to improve DTMF recognition so people can just type in > stuff without having to hold the keys down? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/dd12f27d/attachment-0001.html From austad at signal15.com Wed May 6 11:56:51 2009 From: austad at signal15.com (Jay Austad) Date: Wed, 6 May 2009 13:56:51 -0500 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: References: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com> Message-ID: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | austad at signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: > Well it depends.. first off are you doing inband dtmf or RFC2833? > Secondly what SVN rev are you running? > > /b > > On May 6, 2009, at 1:44 PM, Jay Austad wrote: > >> Using the default installation, I've noticed that when I (or someone >> else) calls in on my SIP trunk and keys in an extension, not all of >> the numbers are recognized unless they hold the key down for at least >> 1/2 second to a second. >> >> Is there a way to improve DTMF recognition so people can just type in >> stuff without having to hold the keys down? > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/3ce72c49/attachment.html From msc at freeswitch.org Wed May 6 12:02:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 May 2009 12:02:08 -0700 Subject: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony In-Reply-To: <7b197bef0905061135s62ea2cb4hfd0fb2fc8bacace7@mail.gmail.com> References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com> <7b197bef0905061000v56269cceh73c8b7519662275e@mail.gmail.com> <87f2f3b90905061123h19f32216p235ff71e5e0db44@mail.gmail.com> <7b197bef0905061135s62ea2cb4hfd0fb2fc8bacace7@mail.gmail.com> Message-ID: <87f2f3b90905061202i6e3210e7ybb6abca980019e4c@mail.gmail.com> On Wed, May 6, 2009 at 11:35 AM, Giovanni Maruzzelli wrote: > On Wed, May 6, 2009 at 8:23 PM, Michael Collins > wrote: > > FYI, > > I made a comment on Dave's blog extolling the virtues of FS and I > mentioned > > Skype support. I didn't specifically mention mod_skypiax but I didn't > > specifically mention any mods. > > > > I was suggesting to put mod_skypiax in the > http://www.freeswitch.org/node/182 page, for ourselves > Done! > > BTW: Very nice comment, it sure will attract attention! > Let's hope so. -MC > > -gm > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/511f6380/attachment.html From nik.middleton at noblesolutions.co.uk Wed May 6 13:40:38 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 6 May 2009 21:40:38 +0100 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> References: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com> <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> Message-ID: Hi Jay, Have to say my DTMF works flawlessly on thousands of calls. (SVN trunk from a couple of days ago. We handle around 100,000 calls/day via FS) That said, I've found it depends on your SIP trunk provider. That doesn't mean to say there isn't a problem; it's just that I haven't come across it. Know it's not helpful, but there you go. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jay Austad Sent: 06 May 2009 19:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition flaky I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | austad at signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote: Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090506/bd112cb3/attachment-0001.html From exyeechen at hotmail.com Wed May 6 17:58:23 2009 From: exyeechen at hotmail.com (chenexyee) Date: Thu, 7 May 2009 08:58:23 +0800 Subject: [Freeswitch-users] Busy tone and text message configuration In-Reply-To: <4A0194FE.9030609@freeswitch.org> References: <4A0194FE.9030609@freeswitch.org> Message-ID: > Date: Wed, 6 May 2009 09:47:42 -0400 > From: intralanman at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Busy tone and text message configuration > > chenexyee wrote: > > > > 1. user A is in conversation with user B, and at this time, a incoming > > call from user C comes to A, in this case, I want freeswitch to play > > busytone to C, how to configure? > you could use the limit app (mod_limit) to limit A's number of calls to > 1, then play the busy sound with tone_stream or return a 486 to the > caller in the failover extension > > I would make a clarification that it is not my purpose to limit A's number of calls. I just like FS to play the busy sound to any caller who is calling the busy callee(for instance the callee is in conversation or originating a call). someone suggested me to done this by javascript, is it possible?or is there any more simpler solution? Thanks for the detail description. > > 2. I'd like freeswitch to relay text message(use sip),as below scenario: > DISCLAIMER: i'm speaking purely theory here as i've never tried to do this. > but you could probably write something to listen via esl for the > incoming event, then in your program, send that event to the user you > want it relayed to. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________ ?MClub?Messenger????????? http://club.msn.cn/?from=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/59b9a344/attachment.html From dujinfang at gmail.com Wed May 6 20:41:32 2009 From: dujinfang at gmail.com (seven) Date: Thu, 7 May 2009 11:41:32 +0800 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop><87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com><77308CE88F604444863741D590835B10@saeedlaptop><63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com><18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com> <191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com> <62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop> Message-ID: <2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com> See this: http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote: > Thanks Guys > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Anthony Minessale > Sent: Wednesday, May 06, 2009 3:57 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Inboud Call Queue > > I worked on the patch and added it to trunk rev 13240 > > On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote: > The patch haven't been merged into trunk. It should be as easy as > execute the following command in the FS source code root dir: > > patch < /tmp/the_patch_file_name.diff > > I will post an example on the wiki when I finished, hope be soon. > > On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: > > Hi Seven, > > I am exactly looking for this functionality. > > Please let me know when you are finished with new queue manager app. > I?ll try it in my call center. > > Regarding Patch: is it already part of SVN trunk? If not then could > you help me how to install it, I have no programming background. > > Many Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of seven > Sent: Wednesday, May 06, 2009 4:17 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Inboud Call Queue > > > On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: > > Hi Michael, > > Thanks for a quick reply. > > I would definitely create a test environment, but my question is > that will it work in required way? > > I read that in Mod_fifo agent has to call in queue but I need that > all incoming calls are automatically distributed between available > agents or if all are busy then should go to voicemail. > I'm working on a call center like queue scenario right now, I'm > pretty sure it call automatically distributed to available agents, > but the customer will stay in the queue if all agents are busy by > default. You can bind a key to the channel and play a message > repeatedly to guide the customer to voicemail by press a key. > > Also maybe you need this patch to make the fifo works as desired. > > http://jira.freeswitch.org/browse/MODAPP-272 > > I would join IRC for further assistance. > > Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, May 05, 2009 7:19 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Inboud Call Queue > > > On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed > wrote: > Hi All, > > In an inbound call center scenario is it possible that customers > calls in and calls are distributed between online (who are > registered on FS and in idle state) agents. I saw some on going > discussion on list where it looks that currently it?s not possible > but I am newbie so maybe I didn?t understand it well. If it?s > possible then please give me a start point that how can I implement > it. > > I would start here: > http://wiki.freeswitch.org/wiki/Mod_fifo > > I strongly recommend that you set up a FreeSWITCH server and play > around with it so that you can learn the pros and cons of using the > FIFO queues. It would be best if you could set up a few phones and > set them as FIFO agents and then have someone help you make test > calls so that you can emulate your CC environment. > > Also, you might want to join us on IRC: #freeswitch on > irc.freenode.net - there are several users who've had real world > experience with mod_fifo and they might be in a good position to > answer your questions real-time. > > -MC > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/6283ee3e/attachment-0001.html From stackofstuff.dg at gmail.com Wed May 6 21:20:10 2009 From: stackofstuff.dg at gmail.com (Dave Grootwassink) Date: Thu, 7 May 2009 00:20:10 -0400 Subject: [Freeswitch-users] Amazon EC2 no audio Message-ID: <004901c9cecb$1d58c690$580a53b0$@com> Hello all, Help a n00b out. I have been trying to get an instance of FreeSwitch running up in the Amazon EC2 cloud. I have successfully gotten the package built following the wiki and archives of this list. I can get x-lite to register with the switch and it will set up calls out on my asterlink account. The problem is that there is no audio transfer (so I am assuming RTP problem). The setup: Firewall open ports tcp 0-65535 udp 0-65535 --- I tried so many combinations unsuccessfully, I finally just blasted open everything. In conf/freeswitch.xml (174.129.201.96 is assigned elastic IP address) #set "external_rtp_ip=174.129.201.96"
In conf/autoload_configs/sofia.conf.xml Internal network IP assignment: Name: domU-12-31-39-00-84-B6.compute-1.internal Address: 10.254.139.68 When I setup a call through asterlink, I see this in the system log: Ring SDP: v=0 o=FreeSWITCH 1241648830 1241648831 IN IP4 10.254.139.68 s=FreeSWITCH c=IN IP4 10.254.139.68 t=0 0 m=audio 17654 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I do not know SDP parameters off the top of my head, but I am assuming that it is telling Asterlink to route the RTP to the internal network IP address and not the external one. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/4cafb470/attachment.html From diego.viola at gmail.com Wed May 6 23:11:33 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 7 May 2009 02:11:33 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> Message-ID: <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> Hi guys, It's me again, does anyone knows why this doesn't work? require 'rubygems' require 'eventmachine' require 'ESL' EventMachine.run { con = EventMachine::start_server "127.0.0.1", 8084 do fd = con.to_i esl = ESL::ESLconnection.new(fd) esl.execute('answer') end } But using it with the normal TCPServer works? I'm trying to use ESL with EventMachine, but it doesn't appear to work. Although it does with the normal TCPServer. Thanks, On Sun, May 3, 2009 at 5:43 PM, Diego Viola wrote: > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example > > Added. > > On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: >> Will post some examples on the wiki now :) >> >> Diego >> >> On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: >>> NICE! It works, it works =D >>> >>> require 'socket' >>> require 'ESL' >>> >>> server = TCPServer.new(8084) >>> loop do >>> con = server.accept >>> fd = con.to_i >>> esl = ESL::ESLconnection.new(fd) >>> esl.execute('answer') >>> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') >>> end >>> >>> Thanks everyone :D >>> >>> Diego >>> >>> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >>>> I think its con.fileno in this case? ?Not sure. >>>> /b >>>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >>>> >>>> Yep, it works Guido. >>>> >>>> require 'socket' >>>> >>>> server = TCPServer.new(8084) >>>> loop do >>>> ???????con = server.accept >>>> ???????con.puts "connect\n\n" >>>> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>>> answer\n\n" >>>> ???????con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >>>> end >>>> >>>> Thanks for the tip =D >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > From R.Kloosterman at mtel.nl Thu May 7 00:04:49 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Thu, 7 May 2009 09:04:49 +0200 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> References: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com> <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> Hi Jay, Did you make a wireshark trace yet? You should be able to find out exactly what's going on there, which protocol is used, etc. We've had our share of problems with DTMF over SIP trunks as well. Your problems could also be related to timing issues introduced by multiple gateways. Do you know some details on voicepulse's network? There's lots of variations in implementation out there, unfortunately not always fully compatible. Good luck, Remko Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Jay Austad Verzonden: woensdag 6 mei 2009 20:57 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] DTMF recognition flaky I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet. 2833 is the default right? I haven't changed anything. I'm using voicepulse for my SIP trunks. Is there an option I can add to that definition to force RFC2833? -- jay austad | 612.423.1433 | austad at signal15.com On May 6, 2009, at 1:46 PM, Brian West wrote: Well it depends.. first off are you doing inband dtmf or RFC2833? Secondly what SVN rev are you running? /b On May 6, 2009, at 1:44 PM, Jay Austad wrote: Using the default installation, I've noticed that when I (or someone else) calls in on my SIP trunk and keys in an extension, not all of the numbers are recognized unless they hold the key down for at least 1/2 second to a second. Is there a way to improve DTMF recognition so people can just type in stuff without having to hold the keys down? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/b8cfbaf4/attachment.html From saeedahmad1981 at gmail.com Thu May 7 03:08:44 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 7 May 2009 12:08:44 +0200 Subject: [Freeswitch-users] Inboud Call Queue In-Reply-To: <2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com> References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop><87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com><77308CE88F604444863741D590835B10@saeedlaptop><63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com><18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com><191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com><62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop> <2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com> Message-ID: <7F5652780AE84DFFB371EF62042540A8@saeedlaptop> Thanks Seven I'll try it very soon. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of seven Sent: Thursday, May 07, 2009 5:42 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue See this: http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote: Thanks Guys _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, May 06, 2009 3:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue I worked on the patch and added it to trunk rev 13240 On Wed, May 6, 2009 at 7:53 AM, dujinfang wrote: The patch haven't been merged into trunk. It should be as easy as execute the following command in the FS source code root dir: patch < /tmp/the_patch_file_name.diff I will post an example on the wiki when I finished, hope be soon. On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote: Hi Seven, I am exactly looking for this functionality. Please let me know when you are finished with new queue manager app. I'll try it in my call center. Regarding Patch: is it already part of SVN trunk? If not then could you help me how to install it, I have no programming background. Many Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of seven Sent: Wednesday, May 06, 2009 4:17 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote: Hi Michael, Thanks for a quick reply. I would definitely create a test environment, but my question is that will it work in required way? I read that in Mod_fifo agent has to call in queue but I need that all incoming calls are automatically distributed between available agents or if all are busy then should go to voicemail. I'm working on a call center like queue scenario right now, I'm pretty sure it call automatically distributed to available agents, but the customer will stay in the queue if all agents are busy by default. You can bind a key to the channel and play a message repeatedly to guide the customer to voicemail by press a key. Also maybe you need this patch to make the fifo works as desired. http://jira.freeswitch.org/browse/MODAPP-272 I would join IRC for further assistance. Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 05, 2009 7:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call Queue On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed wrote: Hi All, In an inbound call center scenario is it possible that customers calls in and calls are distributed between online (who are registered on FS and in idle state) agents. I saw some on going discussion on list where it looks that currently it's not possible but I am newbie so maybe I didn't understand it well. If it's possible then please give me a start point that how can I implement it. I would start here: http://wiki.freeswitch.org/wiki/Mod_fifo I strongly recommend that you set up a FreeSWITCH server and play around with it so that you can learn the pros and cons of using the FIFO queues. It would be best if you could set up a few phones and set them as FIFO agents and then have someone help you make test calls so that you can emulate your CC environment. Also, you might want to join us on IRC: #freeswitch on irc.freenode.net - there are several users who've had real world experience with mod_fifo and they might be in a good position to answer your questions real-time. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/579ffbfa/attachment-0001.html From dujinfang at gmail.com Thu May 7 03:12:46 2009 From: dujinfang at gmail.com (seven) Date: Thu, 7 May 2009 18:12:46 +0800 Subject: [Freeswitch-users] voicemail webapi returns 403 Forbidden Message-ID: <8E54B40E-EAA8-4CBD-9947-EDF8922198B1@gmail.com> I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to http://192.168.1.27:8080/api/voicemail/web in the challenge box, no matter I input 1009, 1009 at 192.168.1.27, 1009 at localhost, 1009 at default, I got Error 403. However I use freeswitch/works can get the voicemail interface. but shows 0 messages. However there should be some message. What's wrong of me? it's almost the default config. Thanks. sh-3.2# find storage/ storage/ storage//voicemail storage//voicemail/default storage//voicemail/default/192.168.1.27 storage//voicemail/default/192.168.1.27/1001 storage//voicemail/default/192.168.1.27/1001/ msg_fb314198-7ff8-48a5-957e-cd75885b34ac.wav storage//voicemail/default/192.168.1.27/1008 storage//voicemail/default/192.168.1.27/1009 storage//voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9- b4e4-ba3493bb5dc5.wav sqlite> select * from voicemail_msgs; 1240973763|1241405588|1009|192.168.1.27|9e5decb2-bb4a-4287-b85f- ddb935878944|Extension 1001|1001|inbox|/usr/local/fs-090428/storage/ voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9-b4e4- ba3493bb5dc5.wav|15|save|A_URGENT From jason at jasonjgw.net Thu May 7 03:16:32 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 7 May 2009 20:16:32 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> Message-ID: <20090507101632.GA21616@jdc.jasonjgw.net> Remko Kloosterman wrote: > > > Did you make a wireshark trace yet? You should be able to find out > exactly what's going on there, which protocol is used, etc. We've had > our share of problems with DTMF over SIP trunks as well. I've just discovered that I'm having a similar problem to the one discussed in this thread. Here are the symptoms. 1. If I call FreeSWITCH from my Snom 320 SIP phone, DTMF recognition works perfectly. This is also true if I call a friend's FreeSWITCH system. 2. If I call a certain VoIP provider from the Snom phone, via FreeSWITCH (phone -> FreeSWITCH -> provider) and call the provider's DTMF test, DTMF recognition fails to work. Apparently this provider accepts only RFC-2833, which is what FreeSWITCH should be issuing - I haven't changed the settings in the external profile from the defaults. 3. If I call the same provider's DTMF test from PortAudio and issue the pa dtmf command, the provider recognizes the DTMF traffic correctly. I couldn't find any obvious configuration errors on the phone or in my internal and external Sofia profiles. I'll gladly run tshark if that's the next step to take. I can also try setting in the internal profile, but this shouldn't be necessary, since as the wiki states in documenting this variable, FreeSWITCH should decode and re-encode the RFC2833 data anyway when this is set to false. I'll keep working on this, but in the meantime, suggestions are welcome. From kjv at ken-ton.com Thu May 7 04:44:27 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Thu, 7 May 2009 07:44:27 -0400 Subject: [Freeswitch-users] rtpDir Freeswitch In-Reply-To: References: Message-ID: <9AA08D9E-02A8-4AC5-8BB4-66F4C19C7D7E@ken-ton.com> Roger on the previous post asking the question some time ago. I don't have time to get into this sort of thing, I've got too much on my plate right now as it is. Hopefully someone will take up the challenge. I've cross-posted this to the freeswitch forum. 73 de n2vqm Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On May 6, 2009, at 10:28 PM, ham44865 wrote: > > > Karl, I just remembered > about a few months back, someone logged in here and > asked the same thing and I sent him all > the channel drivers I had, except chan_dstar.c(NEW at the time). > Anyway, now it is all OPEN SOURCE. > > I dont know if he did something with them, but I found his posts > and my posts and pasted them now here. > Maybe he did something with them, I dont know. > > In eithe case, chan_rtpdir.c and chan_dstar.c are posted here. > > Question for you: > > chan_rtpdir.c and chan_dstar.c talk to a radio application > within asterisk so that the audio can go over the air. > > Is there any application inside Freeswitch that we can use > to interface chan_dstar.c and chan_rtpdir.c with ? > > Anyway the posts are 711 and 712 and here they are: > > ========================================== > Great, the more options the better. > > I will send you the Echolink channel driver and the > IRLP channel driver for Asterisk. > For IRLP you could also download Speak-Freely but it is > messy code. > Any questions you may have, let me know. > > 73, > de KI4LKF > > --- In rtpDir at yahoogroups.com, "Gerry Hull" wrote: > > > > Hi All, > > > > rptDir looks pretty cool. I am doing a lot of work witrh > FreeSwitch, an > > open-source softswitch (http://www.freeswitch.org). It is much more > > scaleable and flexible than > > Asterisk (though I use Asterisk also). > > > > I'd like to write IRLP/Echolink conferencing for FreeSwitch. Are > their any > > good documents descibing the protocols, or are my only > alternatives to > > reverse-engineer the existing code? > > > > Of course, my resulting code would be open source. > > > > 73, Gerry W1VE > > sip: gerry at w1ve.com > > ================================================ > > --- In rtpDir at yahoogroups.com, Karl Vesterling wrote: > > > > Roger on the channel drivers. > > > > www.freeswitch.org > > > > > > Best Regards, > > Karl J. Vesterling > > kjv at ... > > 202-461-3231 x0 > > > > On May 6, 2009, at 9:00 PM, ham44865 wrote: > > > > > > > > > > > Is there anything in Freeswitch that resembles > > > "channel drivers". > > > > > > I am sure we can take the channel drivers I have for > > > Asterisk and maybe with a few modifications > > > make them run in Freeswitch. > > > > > > For example: I have > > > chan_rtpdir.c(for IRLP, Echolink), > > > and chan_dstar.c (for D-STAR) > > > > > > They call these things channel drivers. > > > I guess in Freeswitch, you might call them "patches", > > > not sure, but whatever they call them in Freeswitch, > > > we can port them to run there also. > > > > > > I am not familiar with Freeswitch, but it seems > > > a interesting project. > > > > > > I am not endorsing asterisk over Freeswitch. > > > I'd say the more competition the better. > > > I do not use asterisk at all, neither Freeswitch. > > > I just write and develop code for others to run. > > > > > > My main platform is rtpDir(D-STAR, Echolink, IRLP). > > > Let me know if I can help with Freeswitch > > > > > > 73 > > > Scott > > > > > > --- In rtpDir at yahoogroups.com, Karl Vesterling wrote: > > > > > > > > Folks; > > > > > > > > I notice there is a way to patch the rtp-dir into Asterisk, > but what > > > > about Freeswitch? > > > > Has there been any development in that area? > > > > > > > > The reason I ask is because I was running Asterisk for years, > but as > > > > of late last summer switched to Freeswitch to gain stability and > > > > capability over Asterisk. > > > > > > > > Hence, setting up rtpdir and integrating it into the VoIP > network > > > > (we're mostly hams) would imply some sort of Asterisk > installation > > > at > > > > each of the locations. And that's not an option any of us are > likely > > > > to entertain. > > > > > > > > The main problem with asterisk problem with asterisk with > asterisk > > > was > > > > the lengthy echoes that we encountered that we encountered > > > encountered > > > > that we encountered 'click'. > > > > > > > > We like our VoIP solid, secure, and free of problems. > > > > > > > > > > > > Best Regards, > > > > Karl J. Vesterling > > > > kjv@ > > > > 202-461-3231 x0 > > > > > > > > On May 6, 2009, at 6:15 PM, ham44865 wrote: > > > > > > > > > > > > > > > > > > > There will be no need any more for the > > > > > dextra_srv D-STAR gateway owner to create or maintain > > > > > the Berkeley DB database file dextra.db used by dextra_srv D- > STAR > > > > > gateway, > > > > > because direct access to ICOM's Postgres database will > > > > > be used by dextra_srv. > > > > > > > > > > As you know, dextra_srv is an OPEN SOURCE replacement > > > > > to the "closed source" dplus. > > > > > > > > > > 73 > > > > > Scott > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > __._,_.___ > Messages in this topic (5)Reply (via web post) | Start a new topic > Messages | Members > > Change settings via the Web (Yahoo! ID required) > Change settings via email: Switch delivery to Daily Digest | Switch > format to Traditional > Visit Your Group | Yahoo! Groups Terms of Use | Unsubscribe > RECENT ACTIVITY > 12 > New Members > 14 > New Files > Visit Your Group > Give Back > Yahoo! for Good > Get inspired > by a good cause. > Y! Toolbar > Get it Free! > easy 1-click access > to your groups. > Yahoo! Groups > Start a group > in 3 easy steps. > Connect with others. > . > > __,_._,___ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/4c81c41d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/4c81c41d/attachment-0001.bin From mikael at bjerkeland.com Thu May 7 04:55:41 2009 From: mikael at bjerkeland.com (Mikael Aleksander Bjerkeland) Date: Thu, 07 May 2009 13:55:41 +0200 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> Message-ID: <1241697341.7025.105.camel@mikael-xpsm1530> EventMachine is very different to TCPSocket and is definitely not a drop-in replacement. Take a look at FreeSWITCHeR (http://code.rubyists.com/projects/fs/repository) and see how they implemented EventMachine. More info about EventMachine and specifically #start_server is here: http://eventmachine.rubyforge.org/EventMachine.html#M000385 El jue, 07-05-2009 a las 02:11 -0400, Diego Viola escribi?: > Hi guys, > > It's me again, does anyone knows why this doesn't work? > > require 'rubygems' > require 'eventmachine' > require 'ESL' > > EventMachine.run { > con = EventMachine::start_server "127.0.0.1", 8084 do > fd = con.to_i > esl = ESL::ESLconnection.new(fd) > esl.execute('answer') > end > } > > But using it with the normal TCPServer works? I'm trying to use ESL > with EventMachine, but it doesn't appear to work. Although it does > with the normal TCPServer. > > Thanks, > > On Sun, May 3, 2009 at 5:43 PM, Diego Viola wrote: > > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example > > > > Added. > > > > On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: > >> Will post some examples on the wiki now :) > >> > >> Diego > >> > >> On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: > >>> NICE! It works, it works =D > >>> > >>> require 'socket' > >>> require 'ESL' > >>> > >>> server = TCPServer.new(8084) > >>> loop do > >>> con = server.accept > >>> fd = con.to_i > >>> esl = ESL::ESLconnection.new(fd) > >>> esl.execute('answer') > >>> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') > >>> end > >>> > >>> Thanks everyone :D > >>> > >>> Diego > >>> > >>> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: > >>>> I think its con.fileno in this case? Not sure. > >>>> /b > >>>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: > >>>> > >>>> Yep, it works Guido. > >>>> > >>>> require 'socket' > >>>> > >>>> server = TCPServer.new(8084) > >>>> loop do > >>>> con = server.accept > >>>> con.puts "connect\n\n" > >>>> con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > >>>> answer\n\n" > >>>> con.puts "sendmsg\ncall-command: execute\nexecute-app-name: > >>>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" > >>>> end > >>>> > >>>> Thanks for the tip =D > >>>> > >>>> Brian West > >>>> brian at freeswitch.org > >>>> -- Meet us at ClueCon! http://www.cluecon.com > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> > >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu May 7 05:09:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 7 May 2009 07:09:09 -0500 Subject: [Freeswitch-users] voicemail webapi returns 403 Forbidden In-Reply-To: <8E54B40E-EAA8-4CBD-9947-EDF8922198B1@gmail.com> References: <8E54B40E-EAA8-4CBD-9947-EDF8922198B1@gmail.com> Message-ID: <191c3a030905070509v5dd61cdfj56adf4f2bd3885ed@mail.gmail.com> in the inside or On Thu, May 7, 2009 at 5:12 AM, seven wrote: > I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to > > http://192.168.1.27:8080/api/voicemail/web > > in the challenge box, no matter I input 1009, 1009 at 192.168.1.27, > 1009 at localhost, 1009 at default, I got Error 403. > > However I use freeswitch/works can get the voicemail interface. but > shows 0 messages. However there should be some message. > > What's wrong of me? it's almost the default config. Thanks. > > sh-3.2# find storage/ > storage/ > storage//voicemail > storage//voicemail/default > storage//voicemail/default/192.168.1.27 > storage//voicemail/default/192.168.1.27/1001 > storage//voicemail/default/192.168.1.27/1001/ > msg_fb314198-7ff8-48a5-957e-cd75885b34ac.wav > storage//voicemail/default/192.168.1.27/1008 > storage//voicemail/default/192.168.1.27/1009 > storage//voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9- > b4e4-ba3493bb5dc5.wav > > sqlite> select * from voicemail_msgs; > 1240973763|1241405588|1009|192.168.1.27|9e5decb2-bb4a-4287-b85f- > ddb935878944|Extension 1001|1001|inbox|/usr/local/fs-090428/storage/ > voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9-b4e4- > ba3493bb5dc5.wav|15|save|A_URGENT > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/12e4d302/attachment.html From anthony.minessale at gmail.com Thu May 7 05:11:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 7 May 2009 07:11:54 -0500 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <20090507101632.GA21616@jdc.jasonjgw.net> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> Message-ID: <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> you may have a sonus infection try some of the stuff from here under DTMF http://wiki.freeswitch.org/wiki/RTP_Issues On Thu, May 7, 2009 at 5:16 AM, Jason White wrote: > Remko Kloosterman wrote: > > > > > > Did you make a wireshark trace yet? You should be able to find out > > exactly what's going on there, which protocol is used, etc. We've had > > our share of problems with DTMF over SIP trunks as well. > > I've just discovered that I'm having a similar problem to the one discussed > in > this thread. Here are the symptoms. > > 1. If I call FreeSWITCH from my Snom 320 SIP phone, DTMF recognition works > perfectly. This is also true if I call a friend's FreeSWITCH system. > > 2. If I call a certain VoIP provider from the Snom phone, via FreeSWITCH > (phone -> FreeSWITCH -> provider) and call the provider's DTMF test, DTMF > recognition fails to work. Apparently this provider accepts only RFC-2833, > which is what FreeSWITCH should be issuing - I haven't changed the settings > in > the external profile from the defaults. > > 3. If I call the same provider's DTMF test from PortAudio and issue the pa > dtmf command, the provider recognizes the DTMF traffic correctly. > > I couldn't find any obvious configuration errors on the phone or in my > internal and external Sofia profiles. > > I'll gladly run tshark if that's the next step to take. > > I can also try setting in the > internal profile, but this shouldn't be necessary, since as the wiki states > in > documenting this variable, FreeSWITCH should decode and re-encode the > RFC2833 > data anyway when this is set to false. > > I'll keep working on this, but in the meantime, suggestions are welcome. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/92643084/attachment.html From anthony.minessale at gmail.com Thu May 7 05:17:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 7 May 2009 07:17:25 -0500 Subject: [Freeswitch-users] rtpDir Freeswitch In-Reply-To: <9AA08D9E-02A8-4AC5-8BB4-66F4C19C7D7E@ken-ton.com> References: <9AA08D9E-02A8-4AC5-8BB4-66F4C19C7D7E@ken-ton.com> Message-ID: <191c3a030905070517w1281e1a7j8aed8dc26fab4c27@mail.gmail.com> To answer the question, "channel drivers" in FreeSWITCH are called "Endpoint Modules" and there is certainly a way to write them we have nearly a dozen in tree. On Thu, May 7, 2009 at 6:44 AM, Karl Vesterling wrote: > > > Roger on the previous post asking the question some time ago. > I don't have time to get into this sort of thing, I've got too much on my > plate right now as it is. > > Hopefully someone will take up the challenge. > > I've cross-posted this to the freeswitch forum. > > 73 de n2vqm > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On May 6, 2009, at 10:28 PM, ham44865 wrote: > > > > Karl, I just remembered > about a few months back, someone logged in here and > asked the same thing and I sent him all > the channel drivers I had, except chan_dstar.c(NEW at the time). > Anyway, now it is all OPEN SOURCE. > > I dont know if he did something with them, but I found his posts > and my posts and pasted them now here. > Maybe he did something with them, I dont know. > > In eithe case, chan_rtpdir.c and chan_dstar.c are posted here. > > Question for you: > > chan_rtpdir.c and chan_dstar.c talk to a radio application > within asterisk so that the audio can go over the air. > > Is there any application inside Freeswitch that we can use > to interface chan_dstar.c and chan_rtpdir.c with ? > > Anyway the posts are 711 and 712 and here they are: > > ========================================== > Great, the more options the better. > > I will send you the Echolink channel driver and the > IRLP channel driver for Asterisk. > For IRLP you could also download Speak-Freely but it is > messy code. > Any questions you may have, let me know. > > 73, > de KI4LKF > > --- In rtpDir at yahoogroups.com , "Gerry Hull" > wrote: > > > > Hi All, > > > > rptDir looks pretty cool. I am doing a lot of work witrh > FreeSwitch, an > > open-source softswitch (http://www.freeswitch.org). It is much more > > scaleable and flexible than > > Asterisk (though I use Asterisk also). > > > > I'd like to write IRLP/Echolink conferencing for FreeSwitch. Are > their any > > good documents descibing the protocols, or are my only alternatives to > > reverse-engineer the existing code? > > > > Of course, my resulting code would be open source. > > > > 73, Gerry W1VE > > sip: gerry at w1ve.com > > ================================================ > > --- In rtpDir at yahoogroups.com , Karl Vesterling > wrote: > > > > Roger on the channel drivers. > > > > www.freeswitch.org > > > > > > Best Regards, > > Karl J. Vesterling > > kjv at ... > > 202-461-3231 x0 > > > > On May 6, 2009, at 9:00 PM, ham44865 wrote: > > > > > > > > > > > Is there anything in Freeswitch that resembles > > > "channel drivers". > > > > > > I am sure we can take the channel drivers I have for > > > Asterisk and maybe with a few modifications > > > make them run in Freeswitch. > > > > > > For example: I have > > > chan_rtpdir.c(for IRLP, Echolink), > > > and chan_dstar.c (for D-STAR) > > > > > > They call these things channel drivers. > > > I guess in Freeswitch, you might call them "patches", > > > not sure, but whatever they call them in Freeswitch, > > > we can port them to run there also. > > > > > > I am not familiar with Freeswitch, but it seems > > > a interesting project. > > > > > > I am not endorsing asterisk over Freeswitch. > > > I'd say the more competition the better. > > > I do not use asterisk at all, neither Freeswitch. > > > I just write and develop code for others to run. > > > > > > My main platform is rtpDir(D-STAR, Echolink, IRLP). > > > Let me know if I can help with Freeswitch > > > > > > 73 > > > Scott > > > > > > --- In rtpDir at yahoogroups.com , Karl > Vesterling wrote: > > > > > > > > Folks; > > > > > > > > I notice there is a way to patch the rtp-dir into Asterisk, but what > > > > about Freeswitch? > > > > Has there been any development in that area? > > > > > > > > The reason I ask is because I was running Asterisk for years, but as > > > > of late last summer switched to Freeswitch to gain stability and > > > > capability over Asterisk. > > > > > > > > Hence, setting up rtpdir and integrating it into the VoIP network > > > > (we're mostly hams) would imply some sort of Asterisk installation > > > at > > > > each of the locations. And that's not an option any of us are likely > > > > to entertain. > > > > > > > > The main problem with asterisk problem with asterisk with asterisk > > > was > > > > the lengthy echoes that we encountered that we encountered > > > encountered > > > > that we encountered 'click'. > > > > > > > > We like our VoIP solid, secure, and free of problems. > > > > > > > > > > > > Best Regards, > > > > Karl J. Vesterling > > > > kjv@ > > > > 202-461-3231 x0 > > > > > > > > On May 6, 2009, at 6:15 PM, ham44865 wrote: > > > > > > > > > > > > > > > > > > > There will be no need any more for the > > > > > dextra_srv D-STAR gateway owner to create or maintain > > > > > the Berkeley DB database file dextra.db used by dextra_srv D-STAR > > > > > gateway, > > > > > because direct access to ICOM's Postgres database will > > > > > be used by dextra_srv. > > > > > > > > > > As you know, dextra_srv is an OPEN SOURCE replacement > > > > > to the "closed source" dplus. > > > > > > > > > > 73 > > > > > Scott > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > __._,_.___ > Messages in this topic > (5)Reply (via web post) > | Start a new topic > Messages > | Members > [image: Yahoo! Groups] > > Change settings via the Web > (Yahoo! ID required) > Change settings via email: Switch delivery to Daily Digest > | Switch format to Traditional > > Visit Your Group > | Yahoo! Groups Terms of Use | > Unsubscribe > RECENT ACTIVITY > > - 12 > New Members > - 14 > New Files > > Visit Your Group > Give Back > Yahoo! for Good > Get inspired > by a good cause. > Y! Toolbar > Get it Free! > easy 1-click access > to your groups. > Yahoo! Groups > Start a group > in 3 easy steps. > Connect with others. > . > [image: Web Bug from > http://geo.yahoo.com/serv?s=97359714/grpId=21832046/grpspId=1705004763/msgId=1916/stime=1241663845/nc1=1/nc2=2/nc3=3] > > __,_._,___ > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/687bb005/attachment-0001.html From codecomplete at free.fr Thu May 7 05:46:06 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 7 May 2009 05:46:06 -0700 (PDT) Subject: [Freeswitch-users] How to install OpenVox PCI card? Message-ID: <23426138.post@talk.nabble.com> Hello I browsed the wiki + archives of this list, but didn't find an article on how to add a TDM card to Freeswitch. If I've overlooked it, thank you for pointing me to it. I have a one-FXO module OpenVox PCI card. It's correctly detected by Linux CentOS 5: ======== # lspci -v 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 5 I/O ports at a000 [size=256] Memory at e2000000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 ======== I've successfully compiled Freeswitch from SVN and connected to it with an SIP softphone. How do I go about adding this card to Freeswitch to handle incoming calls from the PSTN? Thank you. -- View this message in context: http://www.nabble.com/How-to-install-OpenVox-PCI-card--tp23426138p23426138.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu May 7 06:37:35 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 07 May 2009 15:37:35 +0200 Subject: [Freeswitch-users] Double Re-Register problem Message-ID: <4A02E41F.6000307@gmx.net> Hello, I habe the following problem when re-registering to an external SIP provider during a call which results in immediate call-hangups. - FS re-registers with nonce - 2ms later FS re-registers without nonce - external SIP provider asks for credentials - FS re-registers with nonce - External provider hangs up call I think the external equipment (Huawei) gets his messages into disorder and then hangs up. My question is: How can I force FS to only register once (without nonce)? As said, FS tries to register twice within 2 msecs without receiving an answer in between. FS is on a public IP, so there are no NAT problems expected (I can see that until the re-register takes place, media is passed in both directions). Best regards Peter See log: U 2009/05/07 15:04:37.441636 217.xxx.xxx.190:5080 -> 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK43088Ha5QpZBN. Max-Forwards: 70. From: ;tag=jSFF9XmFZ50pp. To: . Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732262 REGISTER. Contact: . Expires: 0. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Authorization: Digest username="0123456779", realm="provider.de", nonce="4a02dc5ba90f927c74161f89e7550138b93f12cc", cnonce="y8SXDLWpEiyNPQAekEzDTg", algorithm=MD5, uri="sip:sip.provider.de;transport=udp", response="1d44f64eec5a5b38b44e398dea201a08", qop=auth, nc=00000002. Content-Length: 0. . # U 2009/05/07 15:04:37.443395 217.xxx.xxx.190:5080 -> 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK5ct1aDU8mZNyg. Max-Forwards: 70. From: ;tag=t1SNQpUB8cKcK. To: . Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732402 REGISTER. Contact: . Expires: 60. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Content-Length: 0. . # U 2009/05/07 15:04:37.466691 213.xxx.xxx.2:5060 -> 217.xxx.xxx.190:5080 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;branch=z9hG4bK5ct1aDU8mZNyg;rport=5080. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. From: ;tag=t1SNQpUB8cKcK. To: ;tag=702dbe10. CSeq: 114732402 REGISTER. Server: SIP Router. WWW-Authenticate: Digest realm="provider.de",nonce="4a02dd789a25b67f29ba21f65429d13c4bbc2ded",qop="auth". Content-Length: 0. . # U 2009/05/07 15:04:37.467211 217.xxx.xxx.190:5080 -> 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK6NKtc8Bcj8BHc. Max-Forwards: 70. From: ;tag=t1SNQpUB8cKcK. To: . Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732403 REGISTER. Contact: . Expires: 60. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Authorization: Digest username="0123456779", realm="provider.de", nonce="4a02dd789a25b67f29ba21f65429d13c4bbc2ded", cnonce="dbAgB7WqEiyNPQAekEzDTg", algorithm=MD5, uri="sip:sip.provider.de;transport=udp", response="6a55b27caec6b06bd9da707e7b24d82b", qop=auth, nc=00000001. Content-Length: 0. . # U 2009/05/07 15:04:37.470935 213.xxx.xxx.2:5060 -> 217.xxx.xxx.190:5080 BYE sip:gw+XXX_01234 at 217.xxx.xxx.190:5080;transport=udp SIP/2.0. Via: SIP/2.0/UDP 213.xxx.xxx.2:5060;branch=z9hG4bK400739ec3ad9e5bbd7f5edccf. Call-ID: d0e38021-b5a9-122c-3d8d-001e904cc34e. From: ;tag=31de8a21. To: "unknown";tag=K287aS5jveQ9H. CSeq: 1 BYE. Max-Forwards: 70. Content-Length: 0. From brian at freeswitch.org Thu May 7 06:43:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 08:43:57 -0500 Subject: [Freeswitch-users] Double Re-Register problem In-Reply-To: <4A02E41F.6000307@gmx.net> References: <4A02E41F.6000307@gmx.net> Message-ID: <0518577A-AE4D-4FB9-AC00-C4FF8D51E6E5@freeswitch.org> Looks like they are sending you a bye... not quite sure this is our problem. Can you provide a sip trace in pcap format for me to see? /b On May 7, 2009, at 8:37 AM, Peter P GMX wrote: > Hello, > > I habe the following problem when re-registering to an external SIP > provider during a call which results in immediate call-hangups. > - FS re-registers with nonce > - 2ms later FS re-registers without nonce > - external SIP provider asks for credentials > - FS re-registers with nonce > - External provider hangs up call > > I think the external equipment (Huawei) gets his messages into > disorder > and then hangs up. > > My question is: How can I force FS to only register once (without > nonce)? > As said, FS tries to register twice within 2 msecs without receiving > an > answer in between. FS is on a public IP, so there are no NAT problems > expected (I can see that until the re-register takes place, media is > passed in both directions). > > > Best regards > Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/8f127745/attachment.html From nameer.kazzaz at gmail.com Thu May 7 06:50:44 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Thu, 07 May 2009 14:50:44 +0100 Subject: [Freeswitch-users] Sip Register Problem Message-ID: <4A02E734.7020202@gmail.com> Hi all, Please if anybody can help me with a problem I'm having. I have 2 gateways Quintum AX and OneAccess 100D both are setup to register with Freeswitch 1.0.4pre7, when I first switch on the gateways all is well and then after about 5 min sip register starts falling with sip 401. I can send what ever log's traces you need. Thank you. From brian at freeswitch.org Thu May 7 07:29:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 09:29:18 -0500 Subject: [Freeswitch-users] Sip Register Problem In-Reply-To: <4A02E734.7020202@gmail.com> References: <4A02E734.7020202@gmail.com> Message-ID: <4A1FB547-2BB3-404A-A5D2-D3A3BD016D49@freeswitch.org> Sounds like the device doesn't challenge correctly. A sip trace would be nice to see. /b On May 7, 2009, at 8:50 AM, Nameer Kazzaz wrote: > Hi all, > Please if anybody can help me with a problem I'm having. I have 2 > gateways Quintum AX and OneAccess 100D both are setup to register with > Freeswitch 1.0.4pre7, when I first switch on the gateways all is well > and then after about 5 min sip register starts falling with sip 401. I > can send what ever log's traces you need. > > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/6f8486d6/attachment.html From helmut.kuper at ewetel.de Thu May 7 08:04:35 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 07 May 2009 17:04:35 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" Message-ID: <4A02F883.9090507@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today a colleague of mine told me that sometimes calls were disconnected without any obvious reasons. In FS's log I found this: 2009-05-07 15:52:22 [ERR] switch_rtp.c:1656 rtp_common_read() Error: SRTP unprotect failed with code 7 (auth check failed) I scanned my recent FS log files for that message and found that this error accours a few times a week. I use Snom Phones all with G722 and SRTP. Any ideas what this could be caused by? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKAviD4tZeNddg3dwRAsdMAKCXJDpum3gEUZTzQDiUgAQ4NBlE/gCgq4WI bJb0huN3fRPyVHhra1vPyAQ= =5sqH -----END PGP SIGNATURE----- From brian at freeswitch.org Thu May 7 08:09:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 10:09:53 -0500 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <4A02F883.9090507@ewetel.de> References: <4A02F883.9090507@ewetel.de> Message-ID: <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> Were they on hold for a bit? /b On May 7, 2009, at 10:04 AM, Helmut Kuper wrote: > I scanned my recent FS log files for that message and found that this > error accours a few times a week. I use Snom Phones all with G722 and > SRTP. Any ideas what this could be caused by? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/83576258/attachment-0001.html From Prometheus001 at gmx.net Thu May 7 08:27:24 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 07 May 2009 17:27:24 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <4A02F883.9090507@ewetel.de> References: <4A02F883.9090507@ewetel.de> Message-ID: <4A02FDDC.10902@gmx.net> Hello Helmut, I also have problems with my Snom300s and Snom320s and G711 and SRTP. They may be related to this problem, but I am not sure. The phones disconnect the media stream after a while (2..10 minutes) because the Snom media port is blocked all of a sudden. I have opened a bug report at Snom [Ticket#2009040810000131]. Best regards Peter Helmut Kuper schrieb: > Hello, > > > today a colleague of mine told me that sometimes calls were disconnected > without any obvious reasons. In FS's log I found this: > > 2009-05-07 15:52:22 [ERR] switch_rtp.c:1656 rtp_common_read() Error: > SRTP unprotect failed with code 7 (auth check failed) > > I scanned my recent FS log files for that message and found that this > error accours a few times a week. I use Snom Phones all with G722 and > SRTP. Any ideas what this could be caused by? > > regards > helmut > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From asobihoudai at yahoo.com Thu May 7 08:35:46 2009 From: asobihoudai at yahoo.com (Paul) Date: Thu, 7 May 2009 08:35:46 -0700 (PDT) Subject: [Freeswitch-users] FS + encryption Message-ID: <807578.97556.qm@web111313.mail.gq1.yahoo.com> Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. I was just curious if the only way to have true end to end secure communications with FS would have to be a SIP trunk from one FS system to another encrypted SIP system on the other with no POTS/PRI/BRI circuits used in transit. I'm assuming if there's any POTS/BRI/PRI/DSS circuits used in transit, anyone with a lineman's handset could still eavesdrop on any conversations. Is this not the case? Paul From brian at freeswitch.org Thu May 7 08:43:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 10:43:53 -0500 Subject: [Freeswitch-users] FS + encryption In-Reply-To: <807578.97556.qm@web111313.mail.gq1.yahoo.com> References: <807578.97556.qm@web111313.mail.gq1.yahoo.com> Message-ID: Well its not so easy to take a lineman's handset and eavesdrop on a T1/ BRI/PRI/DSS1 circuit.. that takes way more hardware.... but POTS you just need a lineman's handset. But yes true secure will need to be end to end. /b On May 7, 2009, at 10:35 AM, Paul wrote: > > Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. > I was just curious if the only way to have true end to end secure > communications with FS would have to be a SIP trunk from one FS > system to another encrypted SIP system on the other with no POTS/PRI/ > BRI circuits used in transit. I'm assuming if there's any POTS/BRI/ > PRI/DSS circuits used in transit, anyone with a lineman's handset > could still eavesdrop on any conversations. Is this not the case? > > Paul > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/c26b3672/attachment.html From asobihoudai at yahoo.com Thu May 7 08:52:18 2009 From: asobihoudai at yahoo.com (Paul) Date: Thu, 7 May 2009 08:52:18 -0700 (PDT) Subject: [Freeswitch-users] FS + encryption In-Reply-To: References: <807578.97556.qm@web111313.mail.gq1.yahoo.com> Message-ID: <148296.63657.qm@web111310.mail.gq1.yahoo.com> You're right. Digital circuits are not so easy to tap into as opposed to POTS. I'm thinking about this issue because I'm wondering how the US Govt. sets their DRSN lines up to be secure. From what I read, only Raytheon and Telecore have DRSN-use JITC-approved switches. It seems that list may be old since I've heard that they're now running a 50k device DRSN network with Cisco gear/software. I suppose for their hardware switches, they must be installing some really long haul trunks from end to end around the world for security. Thanks Brian. ________________________________ From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Thursday, May 7, 2009 11:43:53 AM Subject: Re: [Freeswitch-users] FS + encryption Well its not so easy to take a lineman's handset and eavesdrop on a T1/BRI/PRI/DSS1 circuit.. that takes way more hardware.... but POTS you just need a lineman's handset. But yes true secure will need to be end to end. /b On May 7, 2009, at 10:35 AM, Paul wrote: Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. I was just curious if the only way to have true end to end secure communications with FS would have to be a SIP trunk from one FS system to another encrypted SIP system on the other with no POTS/PRI/BRI circuits used in transit. I'm assuming if there's any POTS/BRI/PRI/DSS circuits used in transit, anyone with a lineman's handset could still eavesdrop on any conversations. Is this not the case? Paul Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/3154af50/attachment.html From msc at freeswitch.org Thu May 7 09:00:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 May 2009 09:00:05 -0700 Subject: [Freeswitch-users] How to install OpenVox PCI card? In-Reply-To: <23426138.post@talk.nabble.com> References: <23426138.post@talk.nabble.com> Message-ID: <87f2f3b90905070900q23255b37h6ee5cea30f91123e@mail.gmail.com> If I understand correctly the OpenVox card is a zaptel clone, which means you'd need to download the drivers, install them and then do the OpenZAP install. See the OpenZAP page on the wiki. -MC On Thu, May 7, 2009 at 5:46 AM, Fred-145 wrote: > > Hello > > I browsed the wiki + archives of this list, but didn't find an article on > how to add a TDM card to Freeswitch. If I've overlooked it, thank you for > pointing me to it. > > I have a one-FXO module OpenVox PCI card. It's correctly detected by Linux > CentOS 5: > > ======== > # lspci -v > 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX > Modem/ISDN > interface > Subsystem: Unknown device b100:0003 > Flags: bus master, medium devsel, latency 64, IRQ 5 > I/O ports at a000 [size=256] > Memory at e2000000 (32-bit, non-prefetchable) [size=4K] > Capabilities: [40] Power Management version 2 > ======== > > I've successfully compiled Freeswitch from SVN and connected to it with an > SIP softphone. > > How do I go about adding this card to Freeswitch to handle incoming calls > from the PSTN? > > Thank you. > -- > View this message in context: > http://www.nabble.com/How-to-install-OpenVox-PCI-card--tp23426138p23426138.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/4ec264de/attachment.html From diego.viola at gmail.com Thu May 7 09:03:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 7 May 2009 12:03:05 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <1241697341.7025.105.camel@mikael-xpsm1530> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> <1241697341.7025.105.camel@mikael-xpsm1530> Message-ID: <86a32abc0905070903s46a5a726h45f251e668f2deb6@mail.gmail.com> I see, but it should work with ESL too right? Diego On Thu, May 7, 2009 at 7:55 AM, Mikael Aleksander Bjerkeland wrote: > EventMachine is very different to TCPSocket and is definitely not a > drop-in replacement. Take a look at FreeSWITCHeR > (http://code.rubyists.com/projects/fs/repository) and see how they > implemented EventMachine. > > > More info about EventMachine and specifically #start_server is here: > http://eventmachine.rubyforge.org/EventMachine.html#M000385 > > > > > El jue, 07-05-2009 a las 02:11 -0400, Diego Viola escribi?: >> Hi guys, >> >> It's me again, does anyone knows why this doesn't work? >> >> require 'rubygems' >> require 'eventmachine' >> require 'ESL' >> >> EventMachine.run { >> ? ? ? ? con = EventMachine::start_server "127.0.0.1", 8084 do >> ? ? ? ? ? ? ? ? fd = con.to_i >> ? ? ? ? ? ? ? ? esl = ESL::ESLconnection.new(fd) >> ? ? ? ? ? ? ? ? esl.execute('answer') >> ? ? ? ? end >> } >> >> But using it with the normal TCPServer works? I'm trying to use ESL >> with EventMachine, but it doesn't appear to work. Although it does >> with the normal TCPServer. >> >> Thanks, >> >> On Sun, May 3, 2009 at 5:43 PM, Diego Viola wrote: >> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example >> > >> > Added. >> > >> > On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: >> >> Will post some examples on the wiki now :) >> >> >> >> Diego >> >> >> >> On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: >> >>> NICE! It works, it works =D >> >>> >> >>> require 'socket' >> >>> require 'ESL' >> >>> >> >>> server = TCPServer.new(8084) >> >>> loop do >> >>> con = server.accept >> >>> fd = con.to_i >> >>> esl = ESL::ESLconnection.new(fd) >> >>> esl.execute('answer') >> >>> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') >> >>> end >> >>> >> >>> Thanks everyone :D >> >>> >> >>> Diego >> >>> >> >>> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >> >>>> I think its con.fileno in this case? ?Not sure. >> >>>> /b >> >>>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >> >>>> >> >>>> Yep, it works Guido. >> >>>> >> >>>> require 'socket' >> >>>> >> >>>> server = TCPServer.new(8084) >> >>>> loop do >> >>>> ? ? ? ?con = server.accept >> >>>> ? ? ? ?con.puts "connect\n\n" >> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >> >>>> answer\n\n" >> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >> >>>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >> >>>> end >> >>>> >> >>>> Thanks for the tip =D >> >>>> >> >>>> Brian West >> >>>> brian at freeswitch.org >> >>>> -- Meet us at ClueCon! ?http://www.cluecon.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> Freeswitch-users mailing list >> >>>> Freeswitch-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>>> >> >>> >> >> >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dyfet at gnutelephony.org Thu May 7 09:08:52 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Thu, 07 May 2009 12:08:52 -0400 Subject: [Freeswitch-users] FS + encryption In-Reply-To: <807578.97556.qm@web111313.mail.gq1.yahoo.com> References: <807578.97556.qm@web111313.mail.gq1.yahoo.com> Message-ID: <4A030794.4040602@gnutelephony.org> SIP TLS will protect the SIP session information with static keys via a certificate, assuming of course the call is direct between two peers. It will do nothing for the actual voice channel. There is SRTP, which can be used to create a cryptographic context over RTP. However, the key question is how to exchange the keys. If they are exchanged in the SIP session, even TLS SIP, then there are certificates around, and it is possible to acquire a past rtp session that has been intercepted. ZRTP offers a solution for setting up SRTP cryptographic contexts using distributed and self generated keys (much like gnupg or ssh) that are exchanged between the peers over RTP itself, and validated through a fingerprint hash at both ends. It is of course essential to initially validate the keys in a secure network first, but once that is done, a man-in-the-middle in the key exchange process will then stick out like a sore thumb. Furthermore, since each call uses different per-session generated keys, there is no forward knowledge; breaking one call does not allow one to also decrypt all past calls. Paul wrote: > Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. > I was just curious if the only way to have true end to end secure communications with FS would have to be a SIP trunk from one FS system to another encrypted SIP system on the other with no POTS/PRI/BRI circuits used in transit. I'm assuming if there's any POTS/BRI/PRI/DSS circuits used in transit, anyone with a lineman's handset could still eavesdrop on any conversations. Is this not the case? > > Paul > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 177 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/128b9955/attachment.vcf From codecomplete at free.fr Thu May 7 09:26:45 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 7 May 2009 09:26:45 -0700 (PDT) Subject: [Freeswitch-users] How to install OpenVox PCI card? In-Reply-To: <87f2f3b90905070900q23255b37h6ee5cea30f91123e@mail.gmail.com> References: <23426138.post@talk.nabble.com> <87f2f3b90905070900q23255b37h6ee5cea30f91123e@mail.gmail.com> Message-ID: <23430550.post@talk.nabble.com> mercutioviz wrote: > If I understand correctly the OpenVox card is a zaptel clone, which means > you'd need to download the drivers, install them and then do the OpenZAP > install. See the OpenZAP page on the wiki. Thanks for the tip. I'll give it a shot. In the mean time, I'm interested in feedback from people who use a TDM card with Zaptel + OpenZap + FS in production : Is it stable, professional-grade quality? -- View this message in context: http://www.nabble.com/How-to-install-OpenVox-PCI-card--tp23426138p23430550.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu May 7 09:34:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 7 May 2009 09:34:24 -0700 Subject: [Freeswitch-users] How to install OpenVox PCI card? In-Reply-To: <23430550.post@talk.nabble.com> References: <23426138.post@talk.nabble.com> <87f2f3b90905070900q23255b37h6ee5cea30f91123e@mail.gmail.com> <23430550.post@talk.nabble.com> Message-ID: <87f2f3b90905070934i72fbb2f5sa391fcf7dcda3fa@mail.gmail.com> On Thu, May 7, 2009 at 9:26 AM, Fred-145 wrote: > > > mercutioviz wrote: > > If I understand correctly the OpenVox card is a zaptel clone, which means > > you'd need to download the drivers, install them and then do the OpenZAP > > install. See the OpenZAP page on the wiki. > > Thanks for the tip. I'll give it a shot. In the mean time, I'm interested > in > feedback from people who use a TDM card with Zaptel + OpenZap + FS in > production : Is it stable, professional-grade quality? > Yes, there are a number of people who are using TDM in production. If you're just using an FXO card you should be just fine. If you are using PRI then you might want to use ozmod_libpri but that's a whole different discussion. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/0863857c/attachment.html From codecomplete at free.fr Thu May 7 09:44:24 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 7 May 2009 09:44:24 -0700 (PDT) Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <49FEDA25.2050703@mctelefonia.com> References: <23193738.post@talk.nabble.com> <9dc4a1670904230323o5cc7b8a4s5ec563dbbee86eb9@mail.gmail.com> <9dc4a1670904270546u574fb943h232cb4335bd46c2b@mail.gmail.com> <23295672.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <23364535.post@talk.nabble.com> <49FEA513.8020109@mctelefonia.com> <23366596.post@talk.nabble.com> <49FEDA25.2050703@mctelefonia.com> Message-ID: <23430873.post@talk.nabble.com> Antonio Gallo wrote: > Alix cases are like 6/9 ? from their shop site. I think its easy to find > someone who work with aluminium that can make for you custom boxes for > like like 6/20 ? at pcs Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2) allow for a PCI slot, either on top of the mobo, or away from it :-/ I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/) allow this, and if I can get a good price for a case + PSU. Thank you. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu May 7 09:46:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 7 May 2009 11:46:07 -0500 Subject: [Freeswitch-users] FS + encryption In-Reply-To: <4A030794.4040602@gnutelephony.org> References: <807578.97556.qm@web111313.mail.gq1.yahoo.com> <4A030794.4040602@gnutelephony.org> Message-ID: <191c3a030905070946x4c0815acx1be069f93da16b04@mail.gmail.com> Hey David! You should come by to this year's ClueCon! We still have some speaking slots left. On Thu, May 7, 2009 at 11:08 AM, David Sugar wrote: > SIP TLS will protect the SIP session information with static keys via a > certificate, assuming of course the call is direct between two peers. > It will do nothing for the actual voice channel. > > There is SRTP, which can be used to create a cryptographic context over > RTP. However, the key question is how to exchange the keys. If they > are exchanged in the SIP session, even TLS SIP, then there are > certificates around, and it is possible to acquire a past rtp session > that has been intercepted. > > ZRTP offers a solution for setting up SRTP cryptographic contexts using > distributed and self generated keys (much like gnupg or ssh) that are > exchanged between the peers over RTP itself, and validated through a > fingerprint hash at both ends. It is of course essential to initially > validate the keys in a secure network first, but once that is done, a > man-in-the-middle in the key exchange process will then stick out like a > sore thumb. Furthermore, since each call uses different per-session > generated keys, there is no forward knowledge; breaking one call does > not allow one to also decrypt all past calls. > > Paul wrote: > > Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. > > I was just curious if the only way to have true end to end secure > communications with FS would have to be a SIP trunk from one FS system to > another encrypted SIP system on the other with no POTS/PRI/BRI circuits used > in transit. I'm assuming if there's any POTS/BRI/PRI/DSS circuits used in > transit, anyone with a lineman's handset could still eavesdrop on any > conversations. Is this not the case? > > > > Paul > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/ca149967/attachment.html From dyfet at gnutelephony.org Thu May 7 11:21:37 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Thu, 07 May 2009 14:21:37 -0400 Subject: [Freeswitch-users] FS + encryption In-Reply-To: <191c3a030905070946x4c0815acx1be069f93da16b04@mail.gmail.com> References: <807578.97556.qm@web111313.mail.gq1.yahoo.com> <4A030794.4040602@gnutelephony.org> <191c3a030905070946x4c0815acx1be069f93da16b04@mail.gmail.com> Message-ID: <4A0326B1.30709@gnutelephony.org> If I can find funding for travel presently I would. Anthony Minessale wrote: > Hey David! > > You should come by to this year's ClueCon! > We still have some speaking slots left. > > > On Thu, May 7, 2009 at 11:08 AM, David Sugar > wrote: > > SIP TLS will protect the SIP session information with static keys via a > certificate, assuming of course the call is direct between two peers. > It will do nothing for the actual voice channel. > > There is SRTP, which can be used to create a cryptographic context over > RTP. However, the key question is how to exchange the keys. If they > are exchanged in the SIP session, even TLS SIP, then there are > certificates around, and it is possible to acquire a past rtp session > that has been intercepted. > > ZRTP offers a solution for setting up SRTP cryptographic contexts using > distributed and self generated keys (much like gnupg or ssh) that are > exchanged between the peers over RTP itself, and validated through a > fingerprint hash at both ends. It is of course essential to initially > validate the keys in a secure network first, but once that is done, a > man-in-the-middle in the key exchange process will then stick out like a > sore thumb. Furthermore, since each call uses different per-session > generated keys, there is no forward knowledge; breaking one call does > not allow one to also decrypt all past calls. > > Paul wrote: > > Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. > > I was just curious if the only way to have true end to end secure > communications with FS would have to be a SIP trunk from one FS > system to another encrypted SIP system on the other with no > POTS/PRI/BRI circuits used in transit. I'm assuming if there's any > POTS/BRI/PRI/DSS circuits used in transit, anyone with a lineman's > handset could still eavesdrop on any conversations. Is this not the > case? > > > > Paul > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 177 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/7940fb70/attachment.vcf From ribs at acm.org Thu May 7 11:48:36 2009 From: ribs at acm.org (Larry Edelstein) Date: Thu, 7 May 2009 11:48:36 -0700 Subject: [Freeswitch-users] Amazon EC2 no audio In-Reply-To: <004901c9cecb$1d58c690$580a53b0$@com> References: <004901c9cecb$1d58c690$580a53b0$@com> Message-ID: <2021f8b20905071148m74473894kf72441ce1cfdee6f@mail.gmail.com> I had this same problem but eventually overcame it. I modified the docs at http://wiki.freeswitch.org/wiki/Amazon_ec2 accordingly. I think the problem I had was the internal vs. external IP address, as you've alluded to at the bottom of your message. In addition to the changes you've made, I also modified sip_profiles/internal.xml - see the docs. -larry On Wed, May 6, 2009 at 9:20 PM, Dave Grootwassink wrote: > Hello all, > > > > Help a n00b out. I have been trying to get an instance of FreeSwitch > running up in the Amazon EC2 cloud. > > > > I have successfully gotten the package built following the wiki and > archives of this list. > > > > I can get x-lite to register with the switch and it will set up calls out > on my asterlink account. The problem is that there is no audio transfer (so > I am assuming RTP problem). > > > > The setup: > > > > Firewall open ports > > tcp 0-65535 > > udp 0-65535 --- I tried so many combinations unsuccessfully, I > finally just blasted open everything. > > > > > > In conf/freeswitch.xml (174.129.201.96 is assigned elastic IP address) > > > > > > #set "external_rtp_ip=174.129.201.96" > >
> > > >
> > > > > > In conf/autoload_configs/sofia.conf.xml > > > > > > > > > > Internal network IP assignment: > > Name: domU-12-31-39-00-84-B6.compute-1.internal > > Address: 10.254.139.68 > > > > > > > > When I setup a call through asterlink, I see this in the system log: > > > > Ring SDP: > > v=0 > > o=FreeSWITCH 1241648830 1241648831 IN IP4 10.254.139.68 > > s=FreeSWITCH > > c=IN IP4 10.254.139.68 > > t=0 0 > > m=audio 17654 RTP/AVP 0 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > I do not know SDP parameters off the top of my head, but I am assuming that > it is telling Asterlink to route the RTP to the internal network IP address > and not the external one. > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/aba7b7ac/attachment-0001.html From diego.viola at gmail.com Thu May 7 13:11:07 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 7 May 2009 16:11:07 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905070903s46a5a726h45f251e668f2deb6@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> <1241697341.7025.105.camel@mikael-xpsm1530> <86a32abc0905070903s46a5a726h45f251e668f2deb6@mail.gmail.com> Message-ID: <86a32abc0905071311v5ed73f29m87166fa20a6d5271@mail.gmail.com> Ok, this seems to work: require 'rubygems' require 'eventmachine' module CallingCard def post_init send_data "sendmsg\ncall-command: execute\nexecute-app-name: answer\n\n" send_data "sendmsg\ncall-command: execute\nexecute-app-name: playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" end end EventMachine::run { EventMachine::start_server "127.0.0.1", 8084, CallingCard } But what about ESL? :/ Diego On Thu, May 7, 2009 at 12:03 PM, Diego Viola wrote: > I see, but it should work with ESL too right? > > Diego > > On Thu, May 7, 2009 at 7:55 AM, Mikael Aleksander Bjerkeland > wrote: >> EventMachine is very different to TCPSocket and is definitely not a >> drop-in replacement. Take a look at FreeSWITCHeR >> (http://code.rubyists.com/projects/fs/repository) and see how they >> implemented EventMachine. >> >> >> More info about EventMachine and specifically #start_server is here: >> http://eventmachine.rubyforge.org/EventMachine.html#M000385 >> >> >> >> >> El jue, 07-05-2009 a las 02:11 -0400, Diego Viola escribi?: >>> Hi guys, >>> >>> It's me again, does anyone knows why this doesn't work? >>> >>> require 'rubygems' >>> require 'eventmachine' >>> require 'ESL' >>> >>> EventMachine.run { >>> ? ? ? ? con = EventMachine::start_server "127.0.0.1", 8084 do >>> ? ? ? ? ? ? ? ? fd = con.to_i >>> ? ? ? ? ? ? ? ? esl = ESL::ESLconnection.new(fd) >>> ? ? ? ? ? ? ? ? esl.execute('answer') >>> ? ? ? ? end >>> } >>> >>> But using it with the normal TCPServer works? I'm trying to use ESL >>> with EventMachine, but it doesn't appear to work. Although it does >>> with the normal TCPServer. >>> >>> Thanks, >>> >>> On Sun, May 3, 2009 at 5:43 PM, Diego Viola wrote: >>> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example >>> > >>> > Added. >>> > >>> > On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: >>> >> Will post some examples on the wiki now :) >>> >> >>> >> Diego >>> >> >>> >> On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: >>> >>> NICE! It works, it works =D >>> >>> >>> >>> require 'socket' >>> >>> require 'ESL' >>> >>> >>> >>> server = TCPServer.new(8084) >>> >>> loop do >>> >>> con = server.accept >>> >>> fd = con.to_i >>> >>> esl = ESL::ESLconnection.new(fd) >>> >>> esl.execute('answer') >>> >>> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') >>> >>> end >>> >>> >>> >>> Thanks everyone :D >>> >>> >>> >>> Diego >>> >>> >>> >>> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >>> >>>> I think its con.fileno in this case? ?Not sure. >>> >>>> /b >>> >>>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >>> >>>> >>> >>>> Yep, it works Guido. >>> >>>> >>> >>>> require 'socket' >>> >>>> >>> >>>> server = TCPServer.new(8084) >>> >>>> loop do >>> >>>> ? ? ? ?con = server.accept >>> >>>> ? ? ? ?con.puts "connect\n\n" >>> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>> >>>> answer\n\n" >>> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>> >>>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >>> >>>> end >>> >>>> >>> >>>> Thanks for the tip =D >>> >>>> >>> >>>> Brian West >>> >>>> brian at freeswitch.org >>> >>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> _______________________________________________ >>> >>>> Freeswitch-users mailing list >>> >>>> Freeswitch-users at lists.freeswitch.org >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>> >>>> >>> >>>> >>> >>> >>> >> >>> > >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From diego.viola at gmail.com Thu May 7 13:44:35 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 7 May 2009 16:44:35 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905071311v5ed73f29m87166fa20a6d5271@mail.gmail.com> References: <86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com> <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> <1241697341.7025.105.camel@mikael-xpsm1530> <86a32abc0905070903s46a5a726h45f251e668f2deb6@mail.gmail.com> <86a32abc0905071311v5ed73f29m87166fa20a6d5271@mail.gmail.com> Message-ID: <86a32abc0905071344v31da7e63p64311e8db9d27fe0@mail.gmail.com> It seems like EM (EventMachine) can't be used with ESL. 16:41 < diegoviola> thedonvaughn: i see, so ESL itself can't be used with EM? 16:41 < wyhaines> You can't just hand the socket from EM to ESL. 16:42 < thedonvaughn> prolly not 16:42 < thedonvaughn> and if tmm1 doesn't know how, then i'm going to say no :() 16:42 < wyhaines> EM will invoke callbacks on your protocol object when data comes in (receive_data), and goes out (send_data). When the connection is closed (unbind), etc.... 16:43 < wyhaines> To use ESL, you'd have collect data form receive_data into a buffer, and pass that into ESL, and ESL would have to support being used in that way. 16:43 < diegoviola> ok then i will just use FSR 16:43 < wyhaines> However, it looks like ESL wants to control the socket, which won't work because EM is already controlling the socket. That conversation was in #eventmachine. Diego On Thu, May 7, 2009 at 4:11 PM, Diego Viola wrote: > Ok, this seems to work: > > require 'rubygems' > require 'eventmachine' > > module CallingCard > ? ? ? ?def post_init > ? ? ? ? ? ? ? ?send_data "sendmsg\ncall-command: > execute\nexecute-app-name: answer\n\n" > ? ? ? ? ? ? ? ?send_data "sendmsg\ncall-command: > execute\nexecute-app-name: playback\nexecute-app-arg: > tone_stream://%(10000,0,350,440)\n\n" > ? ? ? ?end > end > > EventMachine::run { > ? ? ? ?EventMachine::start_server "127.0.0.1", 8084, CallingCard > } > > But what about ESL? :/ > > Diego > > On Thu, May 7, 2009 at 12:03 PM, Diego Viola wrote: >> I see, but it should work with ESL too right? >> >> Diego >> >> On Thu, May 7, 2009 at 7:55 AM, Mikael Aleksander Bjerkeland >> wrote: >>> EventMachine is very different to TCPSocket and is definitely not a >>> drop-in replacement. Take a look at FreeSWITCHeR >>> (http://code.rubyists.com/projects/fs/repository) and see how they >>> implemented EventMachine. >>> >>> >>> More info about EventMachine and specifically #start_server is here: >>> http://eventmachine.rubyforge.org/EventMachine.html#M000385 >>> >>> >>> >>> >>> El jue, 07-05-2009 a las 02:11 -0400, Diego Viola escribi?: >>>> Hi guys, >>>> >>>> It's me again, does anyone knows why this doesn't work? >>>> >>>> require 'rubygems' >>>> require 'eventmachine' >>>> require 'ESL' >>>> >>>> EventMachine.run { >>>> ? ? ? ? con = EventMachine::start_server "127.0.0.1", 8084 do >>>> ? ? ? ? ? ? ? ? fd = con.to_i >>>> ? ? ? ? ? ? ? ? esl = ESL::ESLconnection.new(fd) >>>> ? ? ? ? ? ? ? ? esl.execute('answer') >>>> ? ? ? ? end >>>> } >>>> >>>> But using it with the normal TCPServer works? I'm trying to use ESL >>>> with EventMachine, but it doesn't appear to work. Although it does >>>> with the normal TCPServer. >>>> >>>> Thanks, >>>> >>>> On Sun, May 3, 2009 at 5:43 PM, Diego Viola wrote: >>>> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example >>>> > >>>> > Added. >>>> > >>>> > On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: >>>> >> Will post some examples on the wiki now :) >>>> >> >>>> >> Diego >>>> >> >>>> >> On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: >>>> >>> NICE! It works, it works =D >>>> >>> >>>> >>> require 'socket' >>>> >>> require 'ESL' >>>> >>> >>>> >>> server = TCPServer.new(8084) >>>> >>> loop do >>>> >>> con = server.accept >>>> >>> fd = con.to_i >>>> >>> esl = ESL::ESLconnection.new(fd) >>>> >>> esl.execute('answer') >>>> >>> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') >>>> >>> end >>>> >>> >>>> >>> Thanks everyone :D >>>> >>> >>>> >>> Diego >>>> >>> >>>> >>> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >>>> >>>> I think its con.fileno in this case? ?Not sure. >>>> >>>> /b >>>> >>>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >>>> >>>> >>>> >>>> Yep, it works Guido. >>>> >>>> >>>> >>>> require 'socket' >>>> >>>> >>>> >>>> server = TCPServer.new(8084) >>>> >>>> loop do >>>> >>>> ? ? ? ?con = server.accept >>>> >>>> ? ? ? ?con.puts "connect\n\n" >>>> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>>> >>>> answer\n\n" >>>> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>>> >>>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >>>> >>>> end >>>> >>>> >>>> >>>> Thanks for the tip =D >>>> >>>> >>>> >>>> Brian West >>>> >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> Freeswitch-users mailing list >>>> >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> >>> >>>> >> >>>> > >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > From simon at airg.com Thu May 7 13:47:26 2009 From: simon at airg.com (Simon Tang) Date: Thu, 7 May 2009 13:47:26 -0700 Subject: [Freeswitch-users] Originate - using own uuid Message-ID: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> Hi there, I know there's a "feature" that allows us to create a uuid and then use that uuid to do an originate (api create_uuid). Is there any badness if I don't do the "create_uuid" and just do the originate using my own uuid. I experimented and made the following originate call: api originate {origination_caller_id_number=16041234567,originate_timeout=60,originati on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park ...and this works, but just want to make sure if there's any badness of doing so...that is, if I can guarantee that the origination_uuid I will be passing in *WILL* be unique. From brian at freeswitch.org Thu May 7 13:50:49 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 15:50:49 -0500 Subject: [Freeswitch-users] Originate - using own uuid In-Reply-To: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> Message-ID: I would use real UUID's not something as simple as abcdefg cuz FS will hang up on you if you collide. /b On May 7, 2009, at 3:47 PM, Simon Tang wrote: > Hi there, > > I know there's a "feature" that allows us to create a uuid and then > use > that uuid to do an originate (api create_uuid). Is there any > badness if > I don't do the "create_uuid" and just do the originate using my own > uuid. > > I experimented and made the following originate call: > > api originate > {origination_caller_id_number > =16041234567,originate_timeout=60,originati > on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park > > ...and this works, but just want to make sure if there's any badness > of > doing so...that is, if I can guarantee that the origination_uuid I > will > be passing in *WILL* be unique. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/609c5b0e/attachment.html From andrew at hijacked.us Thu May 7 14:02:53 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 7 May 2009 17:02:53 -0400 Subject: [Freeswitch-users] Originate - using own uuid In-Reply-To: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> Message-ID: <20090507210252.GA10958@hijacked.us> On Thu, May 07, 2009 at 01:47:26PM -0700, Simon Tang wrote: > Hi there, > > I know there's a "feature" that allows us to create a uuid and then use > that uuid to do an originate (api create_uuid). Is there any badness if > I don't do the "create_uuid" and just do the originate using my own > uuid. > > I experimented and made the following originate call: > > api originate > {origination_caller_id_number=16041234567,originate_timeout=60,originati > on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park > > ...and this works, but just want to make sure if there's any badness of > doing so...that is, if I can guarantee that the origination_uuid I will > be passing in *WILL* be unique. You also need to be sure that FreeSWITCH isn't going to use it at any point in the past, present or future ;). It's probably better to use create_uuid if at all possible. Andrew From brian at freeswitch.org Thu May 7 14:08:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 16:08:35 -0500 Subject: [Freeswitch-users] Originate - using own uuid In-Reply-To: <20090507210252.GA10958@hijacked.us> References: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> <20090507210252.GA10958@hijacked.us> Message-ID: <751CF539-BD58-4BF6-801D-6C30F9379E19@freeswitch.org> Yah kinda gets messy when trying to track down something if the same uuid gets used over and over in a short period of time. :P /b On May 7, 2009, at 4:02 PM, Andrew Thompson wrote: > You also need to be sure that FreeSWITCH isn't going to use it at any > point in the past, present or future ;). It's probably better to use > create_uuid if at all possible. > > Andrew Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/c750778d/attachment-0001.html From simon at airg.com Thu May 7 14:18:07 2009 From: simon at airg.com (Simon Tang) Date: Thu, 7 May 2009 14:18:07 -0700 Subject: [Freeswitch-users] Originate - using own uuid In-Reply-To: References: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> Message-ID: <872970CF4A55BF42A5337D570860209F01D65986@HPEXCHVS01.exchange.airg> Abcdefg was an example...I was suggesting that I can guarantee uniqueness forever and if I collide, it's my fault and I take the consequences :) Anyway, I will use create_uuid...I was exploring this option before because I thought I couldn't use create_uuid because I made some wrong assumptions. I was assuming that I wouldn't get the response for the create_uuid because I would have applied a Unique-ID filter for the existing session prior to calling create_uuid, which in turn, would filter out the uuid response. But this assumption was incorrect because the UUID *is* the response to the "api create_uuid" call and won't be filtered and is only visible on the socket it was called on (and not others). From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: May 7, 2009 1:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate - using own uuid I would use real UUID's not something as simple as abcdefg cuz FS will hang up on you if you collide. /b On May 7, 2009, at 3:47 PM, Simon Tang wrote: Hi there, I know there's a "feature" that allows us to create a uuid and then use that uuid to do an originate (api create_uuid). Is there any badness if I don't do the "create_uuid" and just do the originate using my own uuid. I experimented and made the following originate call: api originate {origination_caller_id_number=16041234567,originate_timeout=60,originati on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park ...and this works, but just want to make sure if there's any badness of doing so...that is, if I can guarantee that the origination_uuid I will be passing in *WILL* be unique. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/3d0b73a5/attachment.html From brian at freeswitch.org Thu May 7 14:26:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 16:26:04 -0500 Subject: [Freeswitch-users] Originate - using own uuid In-Reply-To: <872970CF4A55BF42A5337D570860209F01D65986@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> <872970CF4A55BF42A5337D570860209F01D65986@HPEXCHVS01.exchange.airg> Message-ID: <4CD7CDFB-B157-4E3B-8730-2867567D1062@freeswitch.org> How about something like this : {origination_caller_id_number =16041234567,originate_timeout=60,origination_uuid=$ {create_uuid()}}sofia/gateway/icall/16041234567 &park You can call an api with ${apiname()} and the results will go into its place. /b On May 7, 2009, at 4:18 PM, Simon Tang wrote: > {origination_caller_id_number > =16041234567,originate_timeout=60,originati > on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/ca1babb8/attachment.html From brian at freeswitch.org Thu May 7 14:52:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 16:52:28 -0500 Subject: [Freeswitch-users] Double Re-Register problem In-Reply-To: <4A02E41F.6000307@gmx.net> References: <4A02E41F.6000307@gmx.net> Message-ID: Also fix your evil mail server cuz I tried tried to email you and it bounced back. You have failed to detail your scenario more... options used .. how you connect things.. every detail counts no matter how small. Then collect it up and put it on Jira. /b On May 7, 2009, at 8:37 AM, Peter P GMX wrote: > Hello, > > I habe the following problem when re-registering to an external SIP > provider during a call which results in immediate call-hangups. > - FS re-registers with nonce > - 2ms later FS re-registers without nonce > - external SIP provider asks for credentials > - FS re-registers with nonce > - External provider hangs up call > > I think the external equipment (Huawei) gets his messages into > disorder > and then hangs up. > > My question is: How can I force FS to only register once (without > nonce)? > As said, FS tries to register twice within 2 msecs without receiving > an > answer in between. FS is on a public IP, so there are no NAT problems > expected (I can see that until the re-register takes place, media is > passed in both directions). > > > Best regards > Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/d9e1ed3e/attachment.html From anthony.minessale at gmail.com Thu May 7 15:45:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 7 May 2009 17:45:19 -0500 Subject: [Freeswitch-users] Originate - using own uuid In-Reply-To: <4CD7CDFB-B157-4E3B-8730-2867567D1062@freeswitch.org> References: <872970CF4A55BF42A5337D570860209F01D6597B@HPEXCHVS01.exchange.airg> <872970CF4A55BF42A5337D570860209F01D65986@HPEXCHVS01.exchange.airg> <4CD7CDFB-B157-4E3B-8730-2867567D1062@freeswitch.org> Message-ID: <191c3a030905071545l54626ea9qd330f356f23d3a52@mail.gmail.com> it's up to you what you do, break it you buy it =D On Thu, May 7, 2009 at 4:26 PM, Brian West wrote: > How about something like this : > {origination_caller_id_number=16041234567,originate_timeout=60,origination_uuid=${create_uuid()}}sofia/gateway/icall/16041234567 > &park > > You can call an api with ${apiname()} and the results will go into its > place. > > /b > > On May 7, 2009, at 4:18 PM, Simon Tang wrote: > > {origination_caller_id_number=16041234567,originate_timeout=60,originati > on_uuid=abcdefg}sofia/gateway/icall/16041234567 &park > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/6977c999/attachment-0001.html From brian at freeswitch.org Thu May 7 15:56:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 17:56:16 -0500 Subject: [Freeswitch-users] Double Re-Register problem In-Reply-To: <4A02E41F.6000307@gmx.net> References: <4A02E41F.6000307@gmx.net> Message-ID: <4C87DA54-15C2-46D1-B79B-81B024F5D2A6@freeswitch.org> Peter, Fix your email server I'm tired of my messages to you getting bounced back. Also upload cleaned up configs without all the comments and DO NOT attach tar.gz files. /b From jason at jasonjgw.net Thu May 7 17:15:56 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 10:15:56 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> Message-ID: <20090508001556.GA13361@jdc.jasonjgw.net> Anthony Minessale wrote: > you may have a sonus infection > > try some of the stuff from here under DTMF > > http://wiki.freeswitch.org/wiki/RTP_Issues Thank you for the suggestion. I tried both the Sonus and Cisco settings in the external profile (running sofia profile external restart reloadxml after making the changes). This didn't help, unfortunately. If I were to make an informed guess, I would expect Cisco equipment to be at the other end, since my ISP has a strong relationship with Cisco. Whatever their solution for carriers is, this is likely to be it, but I could be wrong, of course. I find it interesting that dtmf over PortAudio works, but from the Snom phone it does not. From diego.viola at gmail.com Thu May 7 17:47:42 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 7 May 2009 20:47:42 -0400 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905071344v31da7e63p64311e8db9d27fe0@mail.gmail.com> References: <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> <1241697341.7025.105.camel@mikael-xpsm1530> <86a32abc0905070903s46a5a726h45f251e668f2deb6@mail.gmail.com> <86a32abc0905071311v5ed73f29m87166fa20a6d5271@mail.gmail.com> <86a32abc0905071344v31da7e63p64311e8db9d27fe0@mail.gmail.com> Message-ID: <86a32abc0905071747u3a5369c1vcdc92a58c762a87b@mail.gmail.com> Hi guys, Nevermind with the ESL and EM thing. I was wondering what the getBody() getHeader() and other ESL stuff does behind the scenes, in raw socket, do you know? Thanks, Diego On Thu, May 7, 2009 at 4:44 PM, Diego Viola wrote: > It seems like EM (EventMachine) can't be used with ESL. > > 16:41 < diegoviola> thedonvaughn: i see, so ESL itself can't be used with EM? > 16:41 < wyhaines> You can't just hand the socket from EM to ESL. > 16:42 < thedonvaughn> prolly not > 16:42 < thedonvaughn> and if tmm1 doesn't know how, then i'm going to say no :() > 16:42 < wyhaines> EM will invoke callbacks on your protocol object > when data comes in (receive_data), and goes out (send_data). ?When the > connection is > ? ? ? ? ? ? ? ? ?closed (unbind), etc.... > 16:43 < wyhaines> To use ESL, you'd have collect data form > receive_data into a buffer, and pass that into ESL, and ESL would have > to support being used in > ? ? ? ? ? ? ? ? ?that way. > 16:43 < diegoviola> ok then i will just use FSR > 16:43 < wyhaines> However, it looks like ESL wants to control the > socket, which won't work because EM is already controlling the socket. > > That conversation was in #eventmachine. > > Diego > > On Thu, May 7, 2009 at 4:11 PM, Diego Viola wrote: >> Ok, this seems to work: >> >> require 'rubygems' >> require 'eventmachine' >> >> module CallingCard >> ? ? ? ?def post_init >> ? ? ? ? ? ? ? ?send_data "sendmsg\ncall-command: >> execute\nexecute-app-name: answer\n\n" >> ? ? ? ? ? ? ? ?send_data "sendmsg\ncall-command: >> execute\nexecute-app-name: playback\nexecute-app-arg: >> tone_stream://%(10000,0,350,440)\n\n" >> ? ? ? ?end >> end >> >> EventMachine::run { >> ? ? ? ?EventMachine::start_server "127.0.0.1", 8084, CallingCard >> } >> >> But what about ESL? :/ >> >> Diego >> >> On Thu, May 7, 2009 at 12:03 PM, Diego Viola wrote: >>> I see, but it should work with ESL too right? >>> >>> Diego >>> >>> On Thu, May 7, 2009 at 7:55 AM, Mikael Aleksander Bjerkeland >>> wrote: >>>> EventMachine is very different to TCPSocket and is definitely not a >>>> drop-in replacement. Take a look at FreeSWITCHeR >>>> (http://code.rubyists.com/projects/fs/repository) and see how they >>>> implemented EventMachine. >>>> >>>> >>>> More info about EventMachine and specifically #start_server is here: >>>> http://eventmachine.rubyforge.org/EventMachine.html#M000385 >>>> >>>> >>>> >>>> >>>> El jue, 07-05-2009 a las 02:11 -0400, Diego Viola escribi?: >>>>> Hi guys, >>>>> >>>>> It's me again, does anyone knows why this doesn't work? >>>>> >>>>> require 'rubygems' >>>>> require 'eventmachine' >>>>> require 'ESL' >>>>> >>>>> EventMachine.run { >>>>> ? ? ? ? con = EventMachine::start_server "127.0.0.1", 8084 do >>>>> ? ? ? ? ? ? ? ? fd = con.to_i >>>>> ? ? ? ? ? ? ? ? esl = ESL::ESLconnection.new(fd) >>>>> ? ? ? ? ? ? ? ? esl.execute('answer') >>>>> ? ? ? ? end >>>>> } >>>>> >>>>> But using it with the normal TCPServer works? I'm trying to use ESL >>>>> with EventMachine, but it doesn't appear to work. Although it does >>>>> with the normal TCPServer. >>>>> >>>>> Thanks, >>>>> >>>>> On Sun, May 3, 2009 at 5:43 PM, Diego Viola wrote: >>>>> > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Ruby_Example >>>>> > >>>>> > Added. >>>>> > >>>>> > On Sun, May 3, 2009 at 5:33 PM, Diego Viola wrote: >>>>> >> Will post some examples on the wiki now :) >>>>> >> >>>>> >> Diego >>>>> >> >>>>> >> On Sun, May 3, 2009 at 5:32 PM, Diego Viola wrote: >>>>> >>> NICE! It works, it works =D >>>>> >>> >>>>> >>> require 'socket' >>>>> >>> require 'ESL' >>>>> >>> >>>>> >>> server = TCPServer.new(8084) >>>>> >>> loop do >>>>> >>> con = server.accept >>>>> >>> fd = con.to_i >>>>> >>> esl = ESL::ESLconnection.new(fd) >>>>> >>> esl.execute('answer') >>>>> >>> esl.execute('playback', 'tone_stream://%(10000,0,350,440)') >>>>> >>> end >>>>> >>> >>>>> >>> Thanks everyone :D >>>>> >>> >>>>> >>> Diego >>>>> >>> >>>>> >>> On Sun, May 3, 2009 at 5:29 PM, Brian West wrote: >>>>> >>>> I think its con.fileno in this case? ?Not sure. >>>>> >>>> /b >>>>> >>>> On May 3, 2009, at 4:00 PM, Diego Viola wrote: >>>>> >>>> >>>>> >>>> Yep, it works Guido. >>>>> >>>> >>>>> >>>> require 'socket' >>>>> >>>> >>>>> >>>> server = TCPServer.new(8084) >>>>> >>>> loop do >>>>> >>>> ? ? ? ?con = server.accept >>>>> >>>> ? ? ? ?con.puts "connect\n\n" >>>>> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>>>> >>>> answer\n\n" >>>>> >>>> ? ? ? ?con.puts "sendmsg\ncall-command: execute\nexecute-app-name: >>>>> >>>> playback\nexecute-app-arg: tone_stream://%(10000,0,350,440)\n\n" >>>>> >>>> end >>>>> >>>> >>>>> >>>> Thanks for the tip =D >>>>> >>>> >>>>> >>>> Brian West >>>>> >>>> brian at freeswitch.org >>>>> >>>> -- Meet us at ClueCon! ?http://www.cluecon.com >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> >>>>> >>>> _______________________________________________ >>>>> >>>> Freeswitch-users mailing list >>>>> >>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>>> http://www.freeswitch.org >>>>> >>>> >>>>> >>>> >>>>> >>> >>>>> >> >>>>> > >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >> > From jason at jasonjgw.net Thu May 7 17:53:57 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 10:53:57 +1000 Subject: [Freeswitch-users] Re-2: Ruby and ESL help In-Reply-To: <86a32abc0905071747u3a5369c1vcdc92a58c762a87b@mail.gmail.com> References: <86a32abc0905031432r1f9dae57yb46038e640f584c4@mail.gmail.com> <86a32abc0905031433mdd9628elec5c077d27422322@mail.gmail.com> <86a32abc0905031443s48d157c4wcb6d1376b04c577d@mail.gmail.com> <86a32abc0905062311r2180d295jfe72801b9a1610dd@mail.gmail.com> <1241697341.7025.105.camel@mikael-xpsm1530> <86a32abc0905070903s46a5a726h45f251e668f2deb6@mail.gmail.com> <86a32abc0905071311v5ed73f29m87166fa20a6d5271@mail.gmail.com> <86a32abc0905071344v31da7e63p64311e8db9d27fe0@mail.gmail.com> <86a32abc0905071747u3a5369c1vcdc92a58c762a87b@mail.gmail.com> Message-ID: <20090508005357.GA16427@jdc.jasonjgw.net> Diego Viola wrote: > Hi guys, > > Nevermind with the ESL and EM thing. > > I was wondering what the getBody() getHeader() and other ESL stuff > does behind the scenes, in raw socket, do you know? Why not read the source code? This is free software and open-source, after all. From larclap at yahoo.com Thu May 7 18:13:57 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 7 May 2009 18:13:57 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <23426138.post@talk.nabble.com> References: <23426138.post@talk.nabble.com> Message-ID: <01e601c9cf7a$431d8e60$c958ab20$@com> I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. Why is it that after I launch freeswitch and type in either 'show' or 'status' at the console, it responds with 'Unknown command', but it does accept 'shutdown'? Thanks, Lars From brian at freeswitch.org Thu May 7 18:19:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 20:19:56 -0500 Subject: [Freeswitch-users] Total noob question In-Reply-To: <01e601c9cf7a$431d8e60$c958ab20$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> Message-ID: Its just "status" Also please do not hijack threads.. Click new message and input the address freeswitch-users at lists.freeswitch.org Thanks, Brian On May 7, 2009, at 8:13 PM, Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it > does > accept 'shutdown'? > > Thanks, Lars > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/65d6d3d5/attachment.html From shannon at sacredhearts.us Thu May 7 18:21:42 2009 From: shannon at sacredhearts.us (Shannon) Date: Thu, 7 May 2009 20:21:42 -0500 Subject: [Freeswitch-users] Total noob question In-Reply-To: <01e601c9cf7a$431d8e60$c958ab20$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> Message-ID: <7e2ac3270905071821sab2963ey418b0e1dc40e07a9@mail.gmail.com> Check out the wiki for more info. Wiki.FreeSWITCH.org On Thursday, May 7, 2009, Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? > > Thanks, Lars > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From jason at jasonjgw.net Thu May 7 18:33:25 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 11:33:25 +1000 Subject: [Freeswitch-users] TLS initialization failures In-Reply-To: <20090429042219.GA5801@jdc.jasonjgw.net> References: <20090429042219.GA5801@jdc.jasonjgw.net> Message-ID: <20090508013325.GA20462@jdc.jasonjgw.net> A quick update: I can still reproduce the profile startup failure under revision 13246, but I haven't hit the segfault again. From jason at jasonjgw.net Thu May 7 18:25:15 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 11:25:15 +1000 Subject: [Freeswitch-users] Total noob question In-Reply-To: <01e601c9cf7a$431d8e60$c958ab20$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> Message-ID: <20090508012515.GA18966@jdc.jasonjgw.net> Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? Maybe the mod_commands module wasn't loaded. Check your logs, and try "load mod_commands" from the FreeSWITCH console. That module should have been loaded unless there is something seriously wrong with the build or installation process. From larclap at yahoo.com Thu May 7 19:01:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 7 May 2009 19:01:59 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> Message-ID: <01ed01c9cf80$f8d8c890$ea8a59b0$@com> Sorry for the hijacking, too anxious. I typed only one of the two commands - both Unknown. 2009-05-07 17:52:52 [CONSOLE] switch_core.c:1322 switch_core_init_and_modload() FreeSWITCH Version 1.0.4pre7 (13238M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at fs> show Unknown Command: show freeswitch at fs> status Unknown Command: status From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, May 07, 2009 6:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Its just "status" Also please do not hijack threads.. Click new message and input the address freeswitch-users at lists.freeswitch.org Thanks, Brian On May 7, 2009, at 8:13 PM, Lars Zeb wrote: I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. Why is it that after I launch freeswitch and type in either 'show' or 'status' at the console, it responds with 'Unknown command', but it does accept 'shutdown'? Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/7e696e68/attachment.html From brian at freeswitch.org Thu May 7 19:09:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 7 May 2009 21:09:56 -0500 Subject: [Freeswitch-users] Total noob question In-Reply-To: <01ed01c9cf80$f8d8c890$ea8a59b0$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <01ed01c9cf80$f8d8c890$ea8a59b0$@com> Message-ID: <6B1DF796-8367-46ED-8453-BE529E26D724@freeswitch.org> Weird did you modify anything in modules.conf? /b On May 7, 2009, at 9:01 PM, Lars Zeb wrote: > Sorry for the hijacking, too anxious. > > I typed only one of the two commands ? both Unknown. > > 2009-05-07 17:52:52 [CONSOLE] switch_core.c:1322 > switch_core_init_and_modload() > FreeSWITCH Version 1.0.4pre7 (13238M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > freeswitch at fs> show > Unknown Command: show > freeswitch at fs> status > Unknown Command: status > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090507/94dbdb18/attachment.html From dujinfang at gmail.com Thu May 7 19:41:42 2009 From: dujinfang at gmail.com (seven) Date: Fri, 8 May 2009 10:41:42 +0800 Subject: [Freeswitch-users] voicemail webapi returns 403 Forbidden In-Reply-To: <191c3a030905070509v5dd61cdfj56adf4f2bd3885ed@mail.gmail.com> References: <8E54B40E-EAA8-4CBD-9947-EDF8922198B1@gmail.com> <191c3a030905070509v5dd61cdfj56adf4f2bd3885ed@mail.gmail.com> Message-ID: <1C485B7D-8048-43A8-9420-3A5660645FF4@gmail.com> It works. Thanks. But when I click the flash "Play", it shows undefined. When I click download, it shows another challenge box where I had to input the same user/pass. Is that a problem? On May 7, 2009, at 8:09 PM, Anthony Minessale wrote: > > > in the inside or > > On Thu, May 7, 2009 at 5:12 AM, seven wrote: > I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to > > http://192.168.1.27:8080/api/voicemail/web > > in the challenge box, no matter I input 1009, 1009 at 192.168.1.27, > 1009 at localhost, 1009 at default, I got Error 403. > > However I use freeswitch/works can get the voicemail interface. but > shows 0 messages. However there should be some message. > > What's wrong of me? it's almost the default config. Thanks. > > sh-3.2# find storage/ > storage/ > storage//voicemail > storage//voicemail/default > storage//voicemail/default/192.168.1.27 > storage//voicemail/default/192.168.1.27/1001 > storage//voicemail/default/192.168.1.27/1001/ > msg_fb314198-7ff8-48a5-957e-cd75885b34ac.wav > storage//voicemail/default/192.168.1.27/1008 > storage//voicemail/default/192.168.1.27/1009 > storage//voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9- > b4e4-ba3493bb5dc5.wav > > sqlite> select * from voicemail_msgs; > 1240973763|1241405588|1009|192.168.1.27|9e5decb2-bb4a-4287-b85f- > ddb935878944|Extension 1001|1001|inbox|/usr/local/fs-090428/storage/ > voicemail/default/192.168.1.27/1009/msg_d5c93a31-0765-41c9-b4e4- > ba3493bb5dc5.wav|15|save|A_URGENT > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/fa6ea0f4/attachment-0001.html From dujinfang at gmail.com Fri May 8 00:04:34 2009 From: dujinfang at gmail.com (seven) Date: Fri, 8 May 2009 15:04:34 +0800 Subject: [Freeswitch-users] iLBC codec 97 or 102 Message-ID: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> Hi, We have FreesSWITCH runing on version 11066, which INVITE with iLBC at 30i number 102, however the recent version INVITE with iLBC at 30i number 97. Unfortunately I have a broken sip client only talks with RTP payload 102. So here is the question: 1) What's the difference between 97 and 102? The default value changed since version 11066? 2) Is it configurable? Is it part of sofia-lib or FS? Thank you. From brian at freeswitch.org Fri May 8 00:18:29 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 8 May 2009 02:18:29 -0500 Subject: [Freeswitch-users] iLBC codec 97 or 102 In-Reply-To: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> References: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> Message-ID: <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> Once you're in dynamic range you should compare the codec name not number. /b On May 8, 2009, at 2:04 AM, seven wrote: > Hi, > > We have FreesSWITCH runing on version 11066, which INVITE with > iLBC at 30i number 102, however the recent version INVITE with iLBC at 30i > number 97. Unfortunately I have a broken sip client only talks with > RTP payload 102. So here is the question: > > 1) What's the difference between 97 and 102? The default value changed > since version 11066? Should be no difference. > 2) Is it configurable? Is it part of sofia-lib or FS? Nope shouldn't matter. > > Thank you. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/34ec1ee5/attachment.html From jason at jasonjgw.net Fri May 8 00:19:00 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 17:19:00 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <20090508001556.GA13361@jdc.jasonjgw.net> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> Message-ID: <20090508071900.GA24664@jdc.jasonjgw.net> I've narrowed this problem down. When I call my ISP's DTMF test and issue DTMF from the Snom phone, do_2833() from switch_rtp.c is never called, as evidenced by freeswitch.log. However, if I call a friend's FreeSWITCH box from the phone (via my FreeSWITCH instance), do_2833() is called. It is also called if I use the voicemail extension on my local FreeSWITCH. Finally, if I call my ISP via PortAudio and use the pa dtmf command, do_2833() is called. It's either something in my configuration, or a bug. I'll keep looking. Anyone with ideas is welcome to offer suggestions. From dujinfang at gmail.com Fri May 8 00:50:00 2009 From: dujinfang at gmail.com (seven) Date: Fri, 8 May 2009 15:50:00 +0800 Subject: [Freeswitch-users] iLBC codec 97 or 102 In-Reply-To: <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> References: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> Message-ID: <5B81DC6D-FA37-4727-A071-4B9CBD318A3F@gmail.com> On May 8, 2009, at 3:18 PM, Brian West wrote: > Once you're in dynamic range you should compare the codec name not > number. > > /b > > On May 8, 2009, at 2:04 AM, seven wrote: > >> Hi, >> >> We have FreesSWITCH runing on version 11066, which INVITE with >> iLBC at 30i number 102, however the recent version INVITE with iLBC at 30i >> number 97. Unfortunately I have a broken sip client only talks with >> RTP payload 102. So here is the question: >> >> 1) What's the difference between 97 and 102? The default value >> changed >> since version 11066? > > Should be no difference. > >> 2) Is it configurable? Is it part of sofia-lib or FS? > > Nope shouldn't matter. Thank you. It must be changeable somewhere in the source code, can you help me to find out? I want it send 102 by default. > >> >> Thank you. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/935e3725/attachment.html From brian at freeswitch.org Fri May 8 00:55:03 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 8 May 2009 02:55:03 -0500 Subject: [Freeswitch-users] iLBC codec 97 or 102 In-Reply-To: <5B81DC6D-FA37-4727-A071-4B9CBD318A3F@gmail.com> References: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> <5B81DC6D-FA37-4727-A071-4B9CBD318A3F@gmail.com> Message-ID: No thats not how the dynamic range works. You can change it in mod_ilbc.c but again NOT the right way. Your phone is broken and should be fixed. /b On May 8, 2009, at 2:50 AM, seven wrote: > Thank you. It must be changeable somewhere in the source code, can > you help me to find out? I want it send 102 by default. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/7f13da19/attachment.html From jason at jasonjgw.net Fri May 8 00:55:30 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 17:55:30 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <20090508071900.GA24664@jdc.jasonjgw.net> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> <20090508071900.GA24664@jdc.jasonjgw.net> Message-ID: <20090508075530.GA25748@jdc.jasonjgw.net> Jason White wrote: >It is also called if I use the voicemail > extension on my local FreeSWITCH. Apologies for the nonsense - I meant that switch_rtp_dequeue_dtmf() is called in that case, for DTMF detection. From jason at jasonjgw.net Fri May 8 01:12:03 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 18:12:03 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <20090508075530.GA25748@jdc.jasonjgw.net> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> <20090508071900.GA24664@jdc.jasonjgw.net> <20090508075530.GA25748@jdc.jasonjgw.net> Message-ID: <20090508081203.GA27101@jdc.jasonjgw.net> As a matter of interest, the other end (as reported in its SDP) is BroadWorks. From jason at jasonjgw.net Fri May 8 01:46:31 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 8 May 2009 18:46:31 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <20090508081203.GA27101@jdc.jasonjgw.net> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> <20090508071900.GA24664@jdc.jasonjgw.net> <20090508075530.GA25748@jdc.jasonjgw.net> <20090508081203.GA27101@jdc.jasonjgw.net> Message-ID: <20090508084631.GA28952@jdc.jasonjgw.net> Sorry for all the e-mail... If I turn off the jitter buffer that I had set in the dialplan extension for that provider, DTMF is correctly sent and detected by the other side. I suspect a bug, but maybe this is the desired behaviour. From dujinfang at gmail.com Fri May 8 02:04:30 2009 From: dujinfang at gmail.com (seven) Date: Fri, 8 May 2009 17:04:30 +0800 Subject: [Freeswitch-users] iLBC codec 97 or 102 In-Reply-To: References: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> <5B81DC6D-FA37-4727-A071-4B9CBD318A3F@gmail.com> Message-ID: <3AD726A8-7FC2-4855-8820-DE530DEBCDB6@gmail.com> But there is only two 30ms variants, SWITCH_CODEC_TYPE_AUDIO, 97, "iLBC", "mode=20", SWITCH_CODEC_TYPE_AUDIO, 98, "iLBC", "mode=30", SWITCH_CODEC_TYPE_AUDIO, 102, "iLBC", "mode=30", Why FS INVITE with mode=30 and 97 but not 98/102 as I'm using iLBC at 30i ? v=0 o=FreeSWITCH 6662257026041736756 6335871247713715835 IN IP4 202.102.135.76 s=FreeSWITCH c=IN IP4 202.102.135.76 t=0 0 m=audio 20820 RTP/AVP 97 101 13 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30 On May 8, 2009, at 3:55 PM, Brian West wrote: > No thats not how the dynamic range works. You can change it in > mod_ilbc.c but again NOT the right way. Your phone is broken and > should be fixed. > > /b > > On May 8, 2009, at 2:50 AM, seven wrote: > >> Thank you. It must be changeable somewhere in the source code, can >> you help me to find out? I want it send 102 by default. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/005f37e6/attachment.html From helmut.kuper at ewetel.de Fri May 8 02:29:36 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 08 May 2009 11:29:36 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> Message-ID: <4A03FB80.2070001@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Brian, no, during conversations calls were desconnected. regards Helmut On 07.05.2009 17:09, Brian West wrote: > Were they on hold for a bit? > > /b > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKA/uA4tZeNddg3dwRAky9AJ9qX/PVYcV2wgxtXJz+HhGSMssYkgCeJRQx r1Px8yF2t16XiM8nhG86w4Q= =2GqO -----END PGP SIGNATURE----- From helmut.kuper at ewetel.de Fri May 8 05:01:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 08 May 2009 14:01:05 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <4A03FB80.2070001@ewetel.de> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> <4A03FB80.2070001@ewetel.de> Message-ID: <4A041F01.1010308@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello again, I spent some more time investigating the problem. The scenario seems to be always the same: - -Phone A calls Phone B on PSTN (OpenZAP+Sangoma) via FS. - -Call is established and Phone A has a (G722)SRTP connection to FS. - -After a while (e.g. 14 minutes or 11 minutes or 10 minutes) a call comes in for Phone A from Phone C via PSTN - -FS tries to ring Phone A. Phone A response with "User Busy" - -FS sends "User-Busy" via PSTN to Phone C. - -Right after OpenZAP channel to Phone C is down, FS detects a "Error: SRTP unprotect failed with code 7 (auth check failed)" (This happens only if a total of 9 srtp errors occurs before) - -Connection from Phone A to Phone B is unbridged by FS - -PSTN-connection is disconnected with "NORMAL_CLEARING" I analysed a time period of 5 weeks. By now I detect this problem only for exact the scenario I described above. It seems that the problem applies only to call durations >9-10 minutes and of course SRTP. I'm on actual svn trunk. regards helmut On 08.05.2009 11:29, Helmut Kuper wrote: > Hello Brian, > > no, during conversations calls were desconnected. > > regards > Helmut > > > On 07.05.2009 17:09, Brian West wrote: >> Were they on hold for a bit? > >> /b > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org - -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKBB8B4tZeNddg3dwRAvS+AJ9GXahzbzggXIxylGVAOWchWM95zwCfdOnC nYikUHUxAp3+2HYrX+A0InI= =7VCf -----END PGP SIGNATURE----- From rupa at rupa.com Fri May 8 05:05:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 8 May 2009 07:05:42 -0500 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: <20090508084631.GA28952@jdc.jasonjgw.net> References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> <20090508071900.GA24664@jdc.jasonjgw.net> <20090508075530.GA25748@jdc.jasonjgw.net> <20090508081203.GA27101@jdc.jasonjgw.net> <20090508084631.GA28952@jdc.jasonjgw.net> Message-ID: Sound bugish to me - or at least not desired behavior. I'd suggest opening up a jira (jira.freeswitch.org) with as much documentation as you have so it can be researched and resolved. On Fri, May 8, 2009 at 3:46 AM, Jason White wrote: > Sorry for all the e-mail... > > If I turn off the jitter buffer that I had set in the dialplan extension > for > that provider, DTMF is correctly sent and detected by the other side. > > I suspect a bug, but maybe this is the desired behaviour. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/b2f2f7a7/attachment.html From rupa at rupa.com Fri May 8 05:08:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 8 May 2009 07:08:42 -0500 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> <20090508071900.GA24664@jdc.jasonjgw.net> <20090508075530.GA25748@jdc.jasonjgw.net> <20090508081203.GA27101@jdc.jasonjgw.net> <20090508084631.GA28952@jdc.jasonjgw.net> Message-ID: Also, in general, I believe you want the jitter buffers on the end-point devices only. Not the guy in the middle. So, jitter buffer should be enabled on the phone, not within FS -- unless FS is the endpoint (eg: IVR). On Fri, May 8, 2009 at 7:05 AM, Rupa Schomaker wrote: > Sound bugish to me - or at least not desired behavior. > > I'd suggest opening up a jira (jira.freeswitch.org) with as much > documentation as you have so it can be researched and resolved. > > > On Fri, May 8, 2009 at 3:46 AM, Jason White wrote: > >> Sorry for all the e-mail... >> >> If I turn off the jitter buffer that I had set in the dialplan extension >> for >> that provider, DTMF is correctly sent and detected by the other side. >> >> I suspect a bug, but maybe this is the desired behaviour. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/cb85699d/attachment-0001.html From helmut.kuper at ewetel.de Fri May 8 06:31:33 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 08 May 2009 15:31:33 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <4A041F01.1010308@ewetel.de> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> <4A03FB80.2070001@ewetel.de> <4A041F01.1010308@ewetel.de> Message-ID: <4A043435.30800@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello again, a further and last update for today: Facing snom phones with different FWs with this scenario I found that Snom 370 with FW 8.2.2 has no problems, while for Snom 370 with FW7.3.11 (I guess also below) I was able to reproduce the problem. So it seems to be a Snom problem rather that a FS problem. Have a nice weekend. helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKBDQ14tZeNddg3dwRAiD2AJ96lswBr0O5puba731QGwIdRBwuNgCfYefX oPfTrAoAl/jFvWCNfcGyYjw= =Y3sC -----END PGP SIGNATURE----- From brian at freeswitch.org Fri May 8 06:32:37 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 8 May 2009 08:32:37 -0500 Subject: [Freeswitch-users] iLBC codec 97 or 102 In-Reply-To: <3AD726A8-7FC2-4855-8820-DE530DEBCDB6@gmail.com> References: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> <5B81DC6D-FA37-4727-A071-4B9CBD318A3F@gmail.com> <3AD726A8-7FC2-4855-8820-DE530DEBCDB6@gmail.com> Message-ID: <7587A69C-471A-477B-93E5-B6F8A1211820@freeswitch.org> actually I think we can remove the 102 version that was there for some google talk thing and I don't think we they do that anymore I'll have to test but ... if we invite to you with mod=30 you have to do 30 no exceptions as per the iLBC spec. /b On May 8, 2009, at 4:04 AM, seven wrote: > But there is only two 30ms variants, > > SWITCH_CODEC_TYPE_AUDIO, 97, "iLBC", "mode=20", > SWITCH_CODEC_TYPE_AUDIO, 98, "iLBC", "mode=30", > SWITCH_CODEC_TYPE_AUDIO, 102, "iLBC", "mode=30", > > Why FS INVITE with mode=30 and 97 but not 98/102 as I'm using > iLBC at 30i ? > > > v=0 > o=FreeSWITCH 6662257026041736756 6335871247713715835 IN IP4 > 202.102.135.76 > s=FreeSWITCH > c=IN IP4 202.102.135.76 > t=0 0 > m=audio 20820 RTP/AVP 97 101 13 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:30 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/d23faf29/attachment.html From larclap at yahoo.com Fri May 8 06:54:42 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 8 May 2009 06:54:42 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <20090508012515.GA18966@jdc.jasonjgw.net> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> Message-ID: <026d01c9cfe4$89f41e20$9ddc5a60$@com> In launching freeswitch, I noticed the errors below, plus many more I did not copy here. Also the mod directory had only two modules in it, so I rebuilt from the tarball again. Same results. I did ./configure, make and make install. What have I done wrong? Thanks, Lars 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_console.so **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_logfile.so **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_enum.so **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default template. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template sql. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template example. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template snom. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template linksys. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template asterisk. 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_event_socket.so **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_loopback.so **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_commands.so **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object file: No such file or directory** ... FreeSWITCH Version 1.0.4pre7 (13238M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] fs:/usr/local/freeswitch/mod # l total 104 drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cluechoo-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + ... + + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, May 07, 2009 6:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? Maybe the mod_commands module wasn't loaded. Check your logs, and try "load mod_commands" from the FreeSWITCH console. That module should have been loaded unless there is something seriously wrong with the build or installation process. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From carlos.talbot at gmail.com Fri May 8 07:18:37 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 8 May 2009 10:18:37 -0400 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <4A043435.30800@ewetel.de> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> <4A03FB80.2070001@ewetel.de> <4A041F01.1010308@ewetel.de> <4A043435.30800@ewetel.de> Message-ID: <5800526b0905080718w214d8933rdbb3febea12dd6e4@mail.gmail.com> On Fri, May 8, 2009 at 9:31 AM, Helmut Kuper wrote: > Facing snom phones with different FWs with this scenario I found that > Snom 370 with FW 8.2.2 has no problems, while for Snom 370 with FW7.3.11 > (I guess also below) I was able to reproduce the problem. So it seems to > be a Snom problem rather that a FS problem. > > > Have a nice weekend. > > helmut How did you get 8.2.2 for the 370? I did not find it on the firmware wiki page... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/ad0557c5/attachment.html From dujinfang at gmail.com Fri May 8 07:42:49 2009 From: dujinfang at gmail.com (dujinfang) Date: Fri, 8 May 2009 22:42:49 +0800 Subject: [Freeswitch-users] iLBC codec 97 or 102 In-Reply-To: <7587A69C-471A-477B-93E5-B6F8A1211820@freeswitch.org> References: <650D28B2-A795-4E70-8E32-46E4475ABEEB@gmail.com> <574CBB72-7C89-46E5-9AAC-9A414207DA10@freeswitch.org> <5B81DC6D-FA37-4727-A071-4B9CBD318A3F@gmail.com> <3AD726A8-7FC2-4855-8820-DE530DEBCDB6@gmail.com> <7587A69C-471A-477B-93E5-B6F8A1211820@freeswitch.org> Message-ID: <19F92079-D44E-4F04-8DE4-97C54BD30E61@gmail.com> On May 8, 2009, at 9:32 PM, Brian West wrote: > actually I think we can remove the 102 version that was there for > some google talk thing and I don't think we they do that anymore > I'll have to test but ... if we invite to you with mod=30 you have > to do 30 no exceptions as per the iLBC spec. > sure the client is broken. but it only do the right thing when you invite me with 102 and mod=30. Please don't remove the 102 before I had my client fixed ;) btw, client can invite with 102 to FS without problem. Thank you again. I think I can make the client work by remove the other versions. If we need multi versions in the future,, I'd like to make a patch make it configurable. > /b > > On May 8, 2009, at 4:04 AM, seven wrote: > >> But there is only two 30ms variants, >> >> SWITCH_CODEC_TYPE_AUDIO, 97, "iLBC", "mode=20", >> SWITCH_CODEC_TYPE_AUDIO, 98, "iLBC", "mode=30", >> SWITCH_CODEC_TYPE_AUDIO, 102, "iLBC", "mode=30", >> >> Why FS INVITE with mode=30 and 97 but not 98/102 as I'm using >> iLBC at 30i ? >> >> >> v=0 >> o=FreeSWITCH 6662257026041736756 6335871247713715835 IN IP4 >> 202.102.135.76 >> s=FreeSWITCH >> c=IN IP4 202.102.135.76 >> t=0 0 >> m=audio 20820 RTP/AVP 97 101 13 >> a=rtpmap:97 iLBC/8000 >> a=fmtp:97 mode=30 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:30 > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/469397bb/attachment-0001.html From msc at freeswitch.org Fri May 8 08:29:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 May 2009 08:29:57 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <026d01c9cfe4$89f41e20$9ddc5a60$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> Message-ID: <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> Lars, May I suggest this process: make clean && svn up && ./bootstrap.sh && ./configure && make install It's one step less intensive than starting with a fresh checkout. I just did that on four systems last night that all had a buggy build with cluechoo and file_string and now all four are humming along nicely. -MC On Fri, May 8, 2009 at 6:54 AM, Lars Zeb wrote: > In launching freeswitch, I noticed the errors below, plus many more I did > not copy here. Also the mod directory had only two modules in it, so I > rebuilt from the tarball again. Same results. > > I did ./configure, make and make install. What have I done wrong? > > Thanks, Lars > > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_console.so > **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_logfile.so > **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_enum.so > **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No > such file or directory** > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default > template. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > sql. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > example. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > snom. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > linksys. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > asterisk. > 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_event_socket.so > **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object > file: No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_sofia.so > **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: > No > such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_loopback.so > **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object > file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_commands.so > **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object > file: > No such file or directory** > ... > FreeSWITCH Version 1.0.4pre7 (13238M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > fs:/usr/local/freeswitch/mod # l > total 104 > drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ > drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ > -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* > -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* > > > make[5]: *** [mod_cluechoo.lo] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_cluechoo-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Making install in build > +-------- FreeSWITCH install Complete ----------+ > + FreeSWITCH has been successfully installed. + > + + > ... > + + > +-----------------------------------------------+ > make[1]: *** [install-recursive] Error 1 > make: *** [install] Error 2 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason > White > Sent: Thursday, May 07, 2009 6:25 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Total noob question > > Lars Zeb wrote: > > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > > > Why is it that after I launch freeswitch and type in either 'show' or > > 'status' at the console, it responds with 'Unknown command', but it does > > accept 'shutdown'? > > Maybe the mod_commands module wasn't loaded. Check your logs, and try > "load mod_commands" from the FreeSWITCH console. > > That module should have been loaded unless there is something seriously > wrong > with the build or installation process. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/1a52caee/attachment.html From carlos.talbot at gmail.com Fri May 8 08:31:10 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 8 May 2009 11:31:10 -0400 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <5800526b0905080718w214d8933rdbb3febea12dd6e4@mail.gmail.com> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> <4A03FB80.2070001@ewetel.de> <4A041F01.1010308@ewetel.de> <4A043435.30800@ewetel.de> <5800526b0905080718w214d8933rdbb3febea12dd6e4@mail.gmail.com> Message-ID: <5800526b0905080831n31c745d6r948536633a033960@mail.gmail.com> Nevermind. I figured it out. It's available as beta on the dms server. http://wiki.snom.com/RSS#Release_candidates_7.3.19_and_8.2.2_can_be_downloaded_from_snom.27s_Document_Management_System_.28DMS.29 On Fri, May 8, 2009 at 10:18 AM, Carlos Talbot wrote: > > How did you get 8.2.2 for the 370? I did not find it on the firmware wiki > page... > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/f7b0969e/attachment.html From gregt at cgicommunications.com Fri May 8 09:33:34 2009 From: gregt at cgicommunications.com (Greg Thoen) Date: Fri, 8 May 2009 12:33:34 -0400 Subject: [Freeswitch-users] Stops accepting calls when idle for four minutes Message-ID: <90210A67-07BD-40E4-AD49-6D195C7FDC67@cgicommunications.com> [Just learning Freeswitch, but I really did search this list looking for my answer for awhile] I compiled the latest trunk release from svn two days ago. I have a DID from flowroute.com pointing to it. Added the flowroute IP to acl.conf.domains and added flowroute.xml to conf/sip_profiles/ internal/ with: Start up freeswitch and it works! Dialing into the flowroute number connects, everything works exactly as expected. But if I don't call in for four minutes, it stops taking accepting the call. Freeswitch has not crashed, it seems up and running; but nothing is even logged when I try to call. It's as if it never reaches freeswitch. Shutdown and restart Freeswitch, and again it works. It will continue to work as long as I call in every minute or so. If I wait four minutes and then call in again, the call never makes it to Freeswitch again. Oddly, I have another test machine running Freeswitch 1.03 release and a nearly identical setup with the flowroute number, and it works fine all the time. Anyone think they can point me in the right direction? -- Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/e2a3859e/attachment.html From mike at yes.net.ua Fri May 8 09:49:01 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Fri, 8 May 2009 19:49:01 +0300 Subject: [Freeswitch-users] Stops accepting calls when idle for four minutes In-Reply-To: <90210A67-07BD-40E4-AD49-6D195C7FDC67@cgicommunications.com> References: <90210A67-07BD-40E4-AD49-6D195C7FDC67@cgicommunications.com> Message-ID: <666386713.20090508194901@yes.net.ua> Hello Greg, It's a NAT box issue. Nat bindings expire if no activity. Try adding a: to your gateway params. But to be honest it's flowroute duty to keep a connection alive by sending keepalives. -- Mike Tkachuk From diego.viola at gmail.com Fri May 8 10:20:24 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 8 May 2009 13:20:24 -0400 Subject: [Freeswitch-users] mod_nibblebill question Message-ID: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> Hi everyone, I'm currently developing a calling card application that uses event socket and mod_nibblebill to bill calls. Well, the question is: can mod_nibblebill disconnect a call when the balance is depleted, or when it reaches 0 cash? The wiki says: "Allow for disconnecting or re-routing calls when balance is depleted." But then at the bottom in "future goals" it says: "We don't yet warn or terminate calls when they've reached their limits." So I'm confused, I also don't see any options in how to set that. In case that mod_nibblebill doesn't support this yet, how hard would it be to add this? I'm willing to do it, I guess it's a variable on the XML config and then read that from the mod_nibblebill.c, and do some logic there. Just wondering if anyone had some experience with this, and if the feature is already there. Thanks, Diego From larclap at yahoo.com Fri May 8 10:20:50 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 8 May 2009 10:20:50 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> Message-ID: <02b901c9d001$5622b670$02682350$@com> The mod modules are still not built. I tried with your suggestion. making install mod_cluechoo Compiling mod_cluechoo.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_cluechoo.c -fPIC -DPIC -o .libs/mod_cluechoo.o cc1: warnings being treated as errors mod_cluechoo.c: In function 'go': mod_cluechoo.c:188: error: call to function 'add_D51' without a real prototype mod_cluechoo.c:45: note: 'add_D51' was declared here mod_cluechoo.c:190: error: call to function 'add_sl' without a real prototype mod_cluechoo.c:46: note: 'add_sl' was declared here mod_cluechoo.c: In function 'vgo': mod_cluechoo.c:313: error: call to function 'add_D51' without a real prototype mod_cluechoo.c:45: note: 'add_D51' was declared here mod_cluechoo.c:315: error: call to function 'add_sl' without a real prototype mod_cluechoo.c:46: note: 'add_sl' was declared here mod_cluechoo.c: In function 'add_sl': mod_cluechoo.c:369: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:370: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:370: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:371: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:371: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:373: error: call to function 'add_smoke' without a real prototype mod_cluechoo.c:48: note: 'add_smoke' was declared here mod_cluechoo.c: In function 'add_D51': mod_cluechoo.c:419: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:420: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:422: error: call to function 'add_smoke' without a real prototype mod_cluechoo.c:48: note: 'add_smoke' was declared here make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_cluechoo-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + . + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 08, 2009 8:30 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars, May I suggest this process: make clean && svn up && ./bootstrap.sh && ./configure && make install It's one step less intensive than starting with a fresh checkout. I just did that on four systems last night that all had a buggy build with cluechoo and file_string and now all four are humming along nicely. -MC On Fri, May 8, 2009 at 6:54 AM, Lars Zeb wrote: In launching freeswitch, I noticed the errors below, plus many more I did not copy here. Also the mod directory had only two modules in it, so I rebuilt from the tarball again. Same results. I did ./configure, make and make install. What have I done wrong? Thanks, Lars 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_console.so **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_logfile.so **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_enum.so **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default template. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template sql. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template example. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template snom. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template linksys. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template asterisk. 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_event_socket.so **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_loopback.so **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_commands.so **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object file: No such file or directory** ... FreeSWITCH Version 1.0.4pre7 (13238M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] fs:/usr/local/freeswitch/mod # l total 104 drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cluechoo-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + ... + + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, May 07, 2009 6:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? Maybe the mod_commands module wasn't loaded. Check your logs, and try "load mod_commands" from the FreeSWITCH console. That module should have been loaded unless there is something seriously wrong with the build or installation process. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/438abcb4/attachment-0001.html From msc at freeswitch.org Fri May 8 10:26:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 May 2009 10:26:29 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <02b901c9d001$5622b670$02682350$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> Message-ID: <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> Okay, time for a fresh checkout. If you are on Linux then I recommend that you completely "rm -fr" your freeswitch source directory, then do the quick and dirty install: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Let us know if that works... -MC On Fri, May 8, 2009 at 10:20 AM, Lars Zeb wrote: > The mod modules are still not built. I tried with your suggestion. > > > > making install mod_cluechoo > > Compiling mod_cluechoo.c... > > quiet_libtool: compile: gcc -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 > -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_cluechoo.c -fPIC -DPIC -o > .libs/mod_cluechoo.o > > cc1: warnings being treated as errors > > mod_cluechoo.c: In function ?go?: > > mod_cluechoo.c:188: error: call to function ?add_D51? without a real > prototype > > mod_cluechoo.c:45: note: ?add_D51? was declared here > > mod_cluechoo.c:190: error: call to function ?add_sl? without a real > prototype > > mod_cluechoo.c:46: note: ?add_sl? was declared here > > mod_cluechoo.c: In function ?vgo?: > > mod_cluechoo.c:313: error: call to function ?add_D51? without a real > prototype > > mod_cluechoo.c:45: note: ?add_D51? was declared here > > mod_cluechoo.c:315: error: call to function ?add_sl? without a real > prototype > > mod_cluechoo.c:46: note: ?add_sl? was declared here > > mod_cluechoo.c: In function ?add_sl?: > > mod_cluechoo.c:369: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:370: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:370: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:371: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:371: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:373: error: call to function ?add_smoke? without a real > prototype > > mod_cluechoo.c:48: note: ?add_smoke? was declared here > > mod_cluechoo.c: In function ?add_D51?: > > mod_cluechoo.c:419: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:420: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:422: error: call to function ?add_smoke? without a real > prototype > > mod_cluechoo.c:48: note: ?add_smoke? was declared here > > make[5]: *** [mod_cluechoo.lo] Error 1 > > make[4]: *** [install] Error 1 > > make[3]: *** [mod_cluechoo-install] Error 1 > > make[2]: *** [install-recursive] Error 1 > > Making install in build > > +-------- FreeSWITCH install Complete ----------+ > > + FreeSWITCH has been successfully installed. + > > + + > > + Install sounds: + > > + (uhd-sounds includes hd-sounds, sounds) + > > + (hd-sounds includes sounds) + > > + ? + > > +-----------------------------------------------+ > > make[1]: *** [install-recursive] Error 1 > > make: *** [install] Error 2 > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, May 08, 2009 8:30 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Total noob question > > > > Lars, > > May I suggest this process: > > make clean && svn up && ./bootstrap.sh && ./configure && make install > > It's one step less intensive than starting with a fresh checkout. I just > did that on four systems last night that all had a buggy build with cluechoo > and file_string and now all four are humming along nicely. > > -MC > > On Fri, May 8, 2009 at 6:54 AM, Lars Zeb wrote: > > In launching freeswitch, I noticed the errors below, plus many more I did > not copy here. Also the mod directory had only two modules in it, so I > rebuilt from the tarball again. Same results. > > I did ./configure, make and make install. What have I done wrong? > > Thanks, Lars > > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_console.so > **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_logfile.so > **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_enum.so > **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No > such file or directory** > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default > template. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > sql. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > example. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > snom. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > linksys. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > asterisk. > 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_event_socket.so > **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object > file: No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_sofia.so > **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: > No > such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_loopback.so > **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object > file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_commands.so > **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object > file: > No such file or directory** > ... > > FreeSWITCH Version 1.0.4pre7 (13238M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > fs:/usr/local/freeswitch/mod # l > total 104 > drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ > drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ > -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* > -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* > > > make[5]: *** [mod_cluechoo.lo] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_cluechoo-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Making install in build > +-------- FreeSWITCH install Complete ----------+ > + FreeSWITCH has been successfully installed. + > + + > ... > + + > +-----------------------------------------------+ > make[1]: *** [install-recursive] Error 1 > make: *** [install] Error 2 > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason > White > Sent: Thursday, May 07, 2009 6:25 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Total noob question > > Lars Zeb wrote: > > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > > > Why is it that after I launch freeswitch and type in either 'show' or > > 'status' at the console, it responds with 'Unknown command', but it does > > accept 'shutdown'? > > Maybe the mod_commands module wasn't loaded. Check your logs, and try > "load mod_commands" from the FreeSWITCH console. > > That module should have been loaded unless there is something seriously > wrong > with the build or installation process. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/5c6a1bb3/attachment-0001.html From gregt at cgicommunications.com Fri May 8 11:23:49 2009 From: gregt at cgicommunications.com (Greg Thoen) Date: Fri, 8 May 2009 14:23:49 -0400 Subject: [Freeswitch-users] Stops accepting calls when idle for four minutes In-Reply-To: <666386713.20090508194901@yes.net.ua> References: <90210A67-07BD-40E4-AD49-6D195C7FDC67@cgicommunications.com> <666386713.20090508194901@yes.net.ua> Message-ID: <461764FC-2E50-44F9-ABD4-D4CEBD32FB35@cgicommunications.com> That solved it! Is there any downside to this method of keeping the nat binding alive? -- Greg On May 8, 2009, at 12:49 PM, Mike Tkachuk wrote: > Hello Greg, > > It's a NAT box issue. Nat bindings expire if no activity. > Try adding a: > > > > to your gateway params. > But to be honest it's flowroute duty to keep a connection alive by > sending keepalives. > > -- > Mike Tkachuk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/64dd21a2/attachment.html From diego.viola at gmail.com Fri May 8 11:34:28 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 8 May 2009 14:34:28 -0400 Subject: [Freeswitch-users] mod_nibblebill question In-Reply-To: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> References: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> Message-ID: <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> Oh I see that it has it already :D - Diego On Fri, May 8, 2009 at 1:20 PM, Diego Viola wrote: > Hi everyone, > > I'm currently developing a calling card application that uses event > socket and mod_nibblebill to bill calls. Well, the question is: can > mod_nibblebill disconnect a call when the balance is depleted, or when > it reaches 0 cash? > > The wiki says: > > "Allow for disconnecting or re-routing calls when balance is depleted." > > But then at the bottom in "future goals" it says: > > "We don't yet warn or terminate calls when they've reached their limits." > > So I'm confused, I also don't see any options in how to set that. In > case that mod_nibblebill doesn't support this yet, how hard would it > be to add this? I'm willing to do it, I guess it's a variable on the > XML config and then read that from the mod_nibblebill.c, and do some > logic there. > > Just wondering if anyone had some experience with this, and if the > feature is already there. > > Thanks, > > Diego > From diego.viola at gmail.com Fri May 8 12:08:29 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 8 May 2009 15:08:29 -0400 Subject: [Freeswitch-users] mod_nibblebill question In-Reply-To: <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> References: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> Message-ID: <86a32abc0905081208u76345237q190559b8f0b161d4@mail.gmail.com> I have set these actions: But when it reaches 0 cash it doesn't hangup :(. On Fri, May 8, 2009 at 2:34 PM, Diego Viola wrote: > Oh I see that it has it already :D > > > > > - > > > > > Diego > > On Fri, May 8, 2009 at 1:20 PM, Diego Viola wrote: >> Hi everyone, >> >> I'm currently developing a calling card application that uses event >> socket and mod_nibblebill to bill calls. Well, the question is: can >> mod_nibblebill disconnect a call when the balance is depleted, or when >> it reaches 0 cash? >> >> The wiki says: >> >> "Allow for disconnecting or re-routing calls when balance is depleted." >> >> But then at the bottom in "future goals" it says: >> >> "We don't yet warn or terminate calls when they've reached their limits." >> >> So I'm confused, I also don't see any options in how to set that. In >> case that mod_nibblebill doesn't support this yet, how hard would it >> be to add this? I'm willing to do it, I guess it's a variable on the >> XML config and then read that from the mod_nibblebill.c, and do some >> logic there. >> >> Just wondering if anyone had some experience with this, and if the >> feature is already there. >> >> Thanks, >> >> Diego >> > From fdhege at gmail.com Fri May 8 12:36:10 2009 From: fdhege at gmail.com (Dale) Date: Fri, 8 May 2009 15:36:10 -0400 Subject: [Freeswitch-users] Gateway Outbound Dial Config Problem Message-ID: <9C71AFEB-B3D3-4691-ABA6-E14365024C69@gmail.com> Hello, I am trying to get a new freeswitch installation working and I have it registering to a sip provider just fine and can receive inbound calls but when I try and place a call out through the gateway the switch is rejecting it due to the domain in the to of the INVITE. Here is the INVITE 15:15:19.467351 IP (tos 0x0, ttl 64, id 40045, offset 0, flags [none], proto: UDP (17), length: 1398) 207.4.223.35.5080 > 207.16.137.36.sip: SIP, length: 1370 INVITE sip:16173800299 at 207.16.137.36 SIP/2.0 Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bKcc3a14Zcg8NZp Max-Forwards: 69 From: "FreeSWITCH" ;tag=54FXray2SNaKN To: Call-ID: 6956c6eb-b6a7-122c-5fb1-725bd701687a CSeq: 114786724 INVITE Here is the REGISTER for the same gateway 15:24:27.746984 IP (tos 0x0, ttl 64, id 40075, offset 0, flags [none], proto: UDP (17), length: 661) 207.4.223.35.5080 > 207.16.137.36.sip: SIP, length: 633 REGISTER sip:207.16.137.36;transport=udp SIP/2.0 Via: SIP/2.0/UDP 207.4.223.35:5080;rport;branch=z9hG4bK9H0SmN8jvavXp Max-Forwards: 70 From: ;tag=3jXBNmvUZ3XDe To: Here is my gateway config Here is the bridge from the dialplan I'm currently running FreeSWITCH Version 1.0.trunk (13226M). I hunted around trying to find another option in for the gateway config that would set the to domain to the realm and still send the packet to proxy but could not find one. Any help/hints are welcome. Thanks, -Dale From larclap at yahoo.com Fri May 8 13:46:36 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 8 May 2009 13:46:36 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> Message-ID: <02ee01c9d01e$148fe260$3dafa720$@com> Sorry to report that the results are the same. I executed the 'make' command after the 'wget' and captured stderr and stdout to a file. I uploaded it to http://rapidshare.com/files/230726648/openSuse_11.1_Freeswitch-1.0.4pre7_log .zip.html if it's of any use. The mod_cluechoo errors persist. The only mods in the mod folder are mod_amr.(la & so) and mod-cdr_csv.(la & so). Maybe I should swap another OS. I'm willing to try anything to get freeswitch up. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 08, 2009 10:26 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Okay, time for a fresh checkout. If you are on Linux then I recommend that you completely "rm -fr" your freeswitch source directory, then do the quick and dirty install: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Let us know if that works... -MC On Fri, May 8, 2009 at 10:20 AM, Lars Zeb wrote: The mod modules are still not built. I tried with your suggestion. making install mod_cluechoo Compiling mod_cluechoo.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_cluechoo.c -fPIC -DPIC -o .libs/mod_cluechoo.o cc1: warnings being treated as errors mod_cluechoo.c: In function 'go': mod_cluechoo.c:188: error: call to function 'add_D51' without a real prototype mod_cluechoo.c:45: note: 'add_D51' was declared here mod_cluechoo.c:190: error: call to function 'add_sl' without a real prototype mod_cluechoo.c:46: note: 'add_sl' was declared here mod_cluechoo.c: In function 'vgo': mod_cluechoo.c:313: error: call to function 'add_D51' without a real prototype mod_cluechoo.c:45: note: 'add_D51' was declared here mod_cluechoo.c:315: error: call to function 'add_sl' without a real prototype mod_cluechoo.c:46: note: 'add_sl' was declared here mod_cluechoo.c: In function 'add_sl': mod_cluechoo.c:369: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:370: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:370: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:371: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:371: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:373: error: call to function 'add_smoke' without a real prototype mod_cluechoo.c:48: note: 'add_smoke' was declared here mod_cluechoo.c: In function 'add_D51': mod_cluechoo.c:419: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:420: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:422: error: call to function 'add_smoke' without a real prototype mod_cluechoo.c:48: note: 'add_smoke' was declared here make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_cluechoo-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + . + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 08, 2009 8:30 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars, May I suggest this process: make clean && svn up && ./bootstrap.sh && ./configure && make install It's one step less intensive than starting with a fresh checkout. I just did that on four systems last night that all had a buggy build with cluechoo and file_string and now all four are humming along nicely. -MC On Fri, May 8, 2009 at 6:54 AM, Lars Zeb wrote: In launching freeswitch, I noticed the errors below, plus many more I did not copy here. Also the mod directory had only two modules in it, so I rebuilt from the tarball again. Same results. I did ./configure, make and make install. What have I done wrong? Thanks, Lars 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_console.so **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_logfile.so **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_enum.so **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default template. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template sql. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template example. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template snom. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template linksys. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template asterisk. 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_event_socket.so **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_loopback.so **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_commands.so **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object file: No such file or directory** ... FreeSWITCH Version 1.0.4pre7 (13238M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] fs:/usr/local/freeswitch/mod # l total 104 drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cluechoo-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + ... + + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, May 07, 2009 6:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? Maybe the mod_commands module wasn't loaded. Check your logs, and try "load mod_commands" from the FreeSWITCH console. That module should have been loaded unless there is something seriously wrong with the build or installation process. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/0dda7fbb/attachment-0001.html From msc at freeswitch.org Fri May 8 14:00:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 May 2009 14:00:41 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <02ee01c9d01e$148fe260$3dafa720$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> Message-ID: <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> Well, Tony likes to say that you can compile FreeSWITCH anywhere but he only supports it if the name of the OS rhymes with Mentos. :) We have had much success with CentOS 5.x installs. I can highly recommend it. -MC On Fri, May 8, 2009 at 1:46 PM, Lars Zeb wrote: > Sorry to report that the results are the same. I executed the ?make? > command after the ?wget? and captured stderr and stdout to a file. I > uploaded it to > http://rapidshare.com/files/230726648/openSuse_11.1_Freeswitch-1.0.4pre7_log.zip.htmlif it?s of any use. > > > > The mod_cluechoo errors persist. The only mods in the mod folder are > mod_amr.(la & so) and mod-cdr_csv.(la & so). > > > > Maybe I should swap another OS. I?m willing to try anything to get > freeswitch up. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, May 08, 2009 10:26 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Total noob question > > > > Okay, time for a fresh checkout. If you are on Linux then I recommend that > you completely "rm -fr" your freeswitch source directory, then do the quick > and dirty install: > > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install > > Let us know if that works... > -MC > > On Fri, May 8, 2009 at 10:20 AM, Lars Zeb wrote: > > The mod modules are still not built. I tried with your suggestion. > > > > making install mod_cluechoo > > Compiling mod_cluechoo.c... > > quiet_libtool: compile: gcc -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 > -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_cluechoo.c -fPIC -DPIC -o > .libs/mod_cluechoo.o > > cc1: warnings being treated as errors > > mod_cluechoo.c: In function ?go?: > > mod_cluechoo.c:188: error: call to function ?add_D51? without a real > prototype > > mod_cluechoo.c:45: note: ?add_D51? was declared here > > mod_cluechoo.c:190: error: call to function ?add_sl? without a real > prototype > > mod_cluechoo.c:46: note: ?add_sl? was declared here > > mod_cluechoo.c: In function ?vgo?: > > mod_cluechoo.c:313: error: call to function ?add_D51? without a real > prototype > > mod_cluechoo.c:45: note: ?add_D51? was declared here > > mod_cluechoo.c:315: error: call to function ?add_sl? without a real > prototype > > mod_cluechoo.c:46: note: ?add_sl? was declared here > > mod_cluechoo.c: In function ?add_sl?: > > mod_cluechoo.c:369: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:370: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:370: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:371: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:371: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:373: error: call to function ?add_smoke? without a real > prototype > > mod_cluechoo.c:48: note: ?add_smoke? was declared here > > mod_cluechoo.c: In function ?add_D51?: > > mod_cluechoo.c:419: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:420: error: call to function ?add_man? without a real > prototype > > mod_cluechoo.c:47: note: ?add_man? was declared here > > mod_cluechoo.c:422: error: call to function ?add_smoke? without a real > prototype > > mod_cluechoo.c:48: note: ?add_smoke? was declared here > > make[5]: *** [mod_cluechoo.lo] Error 1 > > make[4]: *** [install] Error 1 > > make[3]: *** [mod_cluechoo-install] Error 1 > > make[2]: *** [install-recursive] Error 1 > > Making install in build > > +-------- FreeSWITCH install Complete ----------+ > > + FreeSWITCH has been successfully installed. + > > + + > > + Install sounds: + > > + (uhd-sounds includes hd-sounds, sounds) + > > + (hd-sounds includes sounds) + > > + ? + > > +-----------------------------------------------+ > > make[1]: *** [install-recursive] Error 1 > > make: *** [install] Error 2 > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, May 08, 2009 8:30 AM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Total noob question > > > > Lars, > > May I suggest this process: > > make clean && svn up && ./bootstrap.sh && ./configure && make install > > It's one step less intensive than starting with a fresh checkout. I just > did that on four systems last night that all had a buggy build with cluechoo > and file_string and now all four are humming along nicely. > > -MC > > On Fri, May 8, 2009 at 6:54 AM, Lars Zeb wrote: > > In launching freeswitch, I noticed the errors below, plus many more I did > not copy here. Also the mod directory had only two modules in it, so I > rebuilt from the tarball again. Same results. > > I did ./configure, make and make install. What have I done wrong? > > Thanks, Lars > > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_console.so > **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_logfile.so > **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_enum.so > **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No > such file or directory** > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default > template. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > sql. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > example. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > snom. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > linksys. > 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template > asterisk. > 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_event_socket.so > **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object > file: No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_sofia.so > **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: > No > such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_loopback.so > **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object > file: > No such file or directory** > 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module > /usr/local/freeswitch/mod/mod_commands.so > **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object > file: > No such file or directory** > ... > > FreeSWITCH Version 1.0.4pre7 (13238M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > fs:/usr/local/freeswitch/mod # l > total 104 > drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ > drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ > -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* > -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* > > > make[5]: *** [mod_cluechoo.lo] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_cluechoo-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Making install in build > +-------- FreeSWITCH install Complete ----------+ > + FreeSWITCH has been successfully installed. + > + + > ... > + + > +-----------------------------------------------+ > make[1]: *** [install-recursive] Error 1 > make: *** [install] Error 2 > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason > White > Sent: Thursday, May 07, 2009 6:25 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Total noob question > > Lars Zeb wrote: > > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > > > Why is it that after I launch freeswitch and type in either 'show' or > > 'status' at the console, it responds with 'Unknown command', but it does > > accept 'shutdown'? > > Maybe the mod_commands module wasn't loaded. Check your logs, and try > "load mod_commands" from the FreeSWITCH console. > > That module should have been loaded unless there is something seriously > wrong > with the build or installation process. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/2de9908c/attachment-0001.html From lon at kickasspixels.com Fri May 8 13:22:00 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 8 May 2009 13:22:00 -0700 Subject: [Freeswitch-users] Install without example configurations Message-ID: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> Is there a way to compile and install without all the example configuration information? A truly clean install out of the gate. Lon From brian at freeswitch.org Fri May 8 14:11:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 8 May 2009 16:11:38 -0500 Subject: [Freeswitch-users] Install without example configurations In-Reply-To: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> References: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> Message-ID: <5DA33B25-34EB-4DBC-8AF7-0546012749B5@freeswitch.org> cd /usr/local/freeswitch/ rm -rf conf but You'll have to do it all by hand its better off to strip down the config once you understand how FreeSWITCH works. /b On May 8, 2009, at 3:22 PM, Lon Baker wrote: > Is there a way to compile and install without all the example > configuration information? > > A truly clean install out of the gate. > > Lon Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/2f67098e/attachment.html From woof at iwoof.org Fri May 8 14:20:09 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 8 May 2009 17:20:09 -0400 Subject: [Freeswitch-users] Install without example configurations In-Reply-To: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> References: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> Message-ID: Woof! On Fri, May 8, 2009 at 4:22 PM, Lon Baker wrote: > A truly clean install out of the gate. We took a different approach. Rather than change what FS comes with, we created an alternate configuration area and point FreeSWITCH to it when we start. Here's the script: http://code.sipfoundry.org/browse/~raw,r=15066/sipXecs/main/sipXfreeSwitch/bin/freeswitch_setup.sh.in It copies all the "good stuff" from the default FS configuration, and skips the dialplan and sip profiles, leaving them blank ready for customization. --Woof! -- --Woof! From msc at freeswitch.org Fri May 8 14:30:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 May 2009 14:30:36 -0700 Subject: [Freeswitch-users] Install without example configurations In-Reply-To: References: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> Message-ID: <87f2f3b90905081430h7265f166m7c5575a6c4ebf8f7@mail.gmail.com> Thanks for the tip! -MC On Fri, May 8, 2009 at 2:20 PM, Andy Spitzer wrote: > Woof! > > On Fri, May 8, 2009 at 4:22 PM, Lon Baker wrote: > > A truly clean install out of the gate. > > We took a different approach. Rather than change what FS comes with, > we created an alternate configuration area and point FreeSWITCH to it > when we start. > > Here's the script: > > http://code.sipfoundry.org/browse/~raw,r=15066/sipXecs/main/sipXfreeSwitch/bin/freeswitch_setup.sh.in > > It copies all the "good stuff" from the default FS configuration, and > skips the dialplan and sip profiles, leaving them blank ready for > customization. > > --Woof! > > > -- > --Woof! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/b81d31ab/attachment.html From msc at freeswitch.org Fri May 8 15:45:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 8 May 2009 15:45:15 -0700 Subject: [Freeswitch-users] Gateway Outbound Dial Config Problem In-Reply-To: <9C71AFEB-B3D3-4691-ABA6-E14365024C69@gmail.com> References: <9C71AFEB-B3D3-4691-ABA6-E14365024C69@gmail.com> Message-ID: <87f2f3b90905081545ga92330fl2abab77f2a8975a8@mail.gmail.com> What does your dialplan entry look like? Also, a debug trace and even a sip trace would be useful. You can use pastepin.freeswitch.org to paste a lot of stuff in a place where everyone can view it and not overwhelm the email list. -MC On Fri, May 8, 2009 at 12:36 PM, Dale wrote: > > Hello, > > I am trying to get a new freeswitch installation working and I have it > registering to a sip provider just fine and can receive inbound calls > but when I try and place a call out through the gateway the switch is > rejecting it due to the domain in the to of the INVITE. > > Here is the INVITE > > 15:15:19.467351 IP (tos 0x0, ttl 64, id 40045, offset 0, flags > [none], proto: UDP (17), length: 1398) 207.4.223.35.5080 > > 207.16.137.36.sip: SIP, length: 1370 > INVITE sip:16173800299 at 207.16.137.36SIP/2.0 > Via: SIP/2.0/UDP 207.4.223.35:5080 > ;rport;branch=z9hG4bKcc3a14Zcg8NZp > Max-Forwards: 69 > From: "FreeSWITCH" 6172384723 at 172.16.12.100;transport=udp>;tag=54FXray2SNaKN > To: > > > Call-ID: 6956c6eb-b6a7-122c-5fb1-725bd701687a > CSeq: 114786724 INVITE > > Here is the REGISTER for the same gateway > > 15:24:27.746984 IP (tos 0x0, ttl 64, id 40075, offset 0, flags > [none], proto: UDP (17), length: 661) 207.4.223.35.5080 > > 207.16.137.36.sip: SIP, length: 633 > REGISTER sip:207.16.137.36;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 207.4.223.35:5080 > ;rport;branch=z9hG4bK9H0SmN8jvavXp > Max-Forwards: 70 > From: ;transport=udp>;tag=3jXBNmvUZ3XDe > To: > > Here is my gateway config > > > > > > > > > > > > > > Here is the bridge from the dialplan > > > > I'm currently running FreeSWITCH Version 1.0.trunk (13226M). > > I hunted around trying to find another option in for the gateway > config that would set the to domain to the realm and still send the > packet to proxy but could not find one. > > Any help/hints are welcome. > > Thanks, > > -Dale > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/79ab4260/attachment.html From fdhege at gmail.com Fri May 8 16:56:10 2009 From: fdhege at gmail.com (Dale) Date: Fri, 8 May 2009 19:56:10 -0400 Subject: [Freeswitch-users] Gateway Outbound Dial Config Problem In-Reply-To: <87f2f3b90905081545ga92330fl2abab77f2a8975a8@mail.gmail.com> References: <9C71AFEB-B3D3-4691-ABA6-E14365024C69@gmail.com> <87f2f3b90905081545ga92330fl2abab77f2a8975a8@mail.gmail.com> Message-ID: <896CEA2A-C82A-4410-AA4A-AF332E5535AE@gmail.com> Thanks for the response. Of course after sending the email I find the problem. The account was not properly setup on the other end to place outbound calls. So everything is working now. Thanks, -Dale On May 8, 2009, at 6:45 PM, Michael Collins wrote: > What does your dialplan entry look like? Also, a debug trace and > even a sip trace would be useful. You can use > pastepin.freeswitch.org to paste a lot of stuff in a place where > everyone can view it and not overwhelm the email list. > > -MC > > On Fri, May 8, 2009 at 12:36 PM, Dale wrote: > > Hello, > > I am trying to get a new freeswitch installation working and I have it > registering to a sip provider just fine and can receive inbound calls > but when I try and place a call out through the gateway the switch is > rejecting it due to the domain in the to of the INVITE. > > Here is the INVITE > > 15:15:19.467351 IP (tos 0x0, ttl 64, id 40045, offset 0, flags > [none], proto: UDP (17), length: 1398) 207.4.223.35.5080 > > 207.16.137.36.sip: SIP, length: 1370 > INVITE sip:16173800299 at 207.16.137.36 SIP/2.0 > Via: SIP/2.0/UDP > 207.4.223.35:5080;rport;branch=z9hG4bKcc3a14Zcg8NZp > Max-Forwards: 69 > From: "FreeSWITCH" 6172384723 at 172.16.12.100;transport=udp>;tag=54FXray2SNaKN > To: > Call-ID: 6956c6eb-b6a7-122c-5fb1-725bd701687a > CSeq: 114786724 INVITE > > Here is the REGISTER for the same gateway > > 15:24:27.746984 IP (tos 0x0, ttl 64, id 40075, offset 0, flags > [none], proto: UDP (17), length: 661) 207.4.223.35.5080 > > 207.16.137.36.sip: SIP, length: 633 > REGISTER sip:207.16.137.36;transport=udp SIP/2.0 > Via: SIP/2.0/UDP > 207.4.223.35:5080;rport;branch=z9hG4bK9H0SmN8jvavXp > Max-Forwards: 70 > From: 6172384723 at 172.16.12.100;transport=udp>;tag=3jXBNmvUZ3XDe > To: > > Here is my gateway config > > > > > > > > > > > > > > Here is the bridge from the dialplan > > > > I'm currently running FreeSWITCH Version 1.0.trunk (13226M). > > I hunted around trying to find another option in for the gateway > config that would set the to domain to the realm and still send the > packet to proxy but could not find one. > > Any help/hints are welcome. > > Thanks, > > -Dale > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090508/f0e07b0d/attachment-0001.html From jason at jasonjgw.net Fri May 8 18:38:48 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 9 May 2009 11:38:48 +1000 Subject: [Freeswitch-users] DTMF recognition flaky In-Reply-To: References: <2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com> <11372C8B9E603F4FACDE6AB18256DEC6017CBD12@srvmtel.office.mtel.nl> <20090507101632.GA21616@jdc.jasonjgw.net> <191c3a030905070511h7c458ae3t54475b82b76f232b@mail.gmail.com> <20090508001556.GA13361@jdc.jasonjgw.net> <20090508071900.GA24664@jdc.jasonjgw.net> <20090508075530.GA25748@jdc.jasonjgw.net> <20090508081203.GA27101@jdc.jasonjgw.net> <20090508084631.GA28952@jdc.jasonjgw.net> Message-ID: <20090509013848.GA20264@jdc.jasonjgw.net> Rupa Schomaker wrote: > Sound bugish to me - or at least not desired behavior. > > I'd suggest opening up a jira (jira.freeswitch.org) with as much > documentation as you have so it can be researched and resolved. If someone could add it to Jira, I'll detail the issue here. The Jira Web interface is a problem for me, and it doesn't seem to allow submissions by e-mail or in other ways. Basically, the problem is that RFC2833 DTMF isn't sent to the other side if a jitterbuffer is set in the dial plan extension for the outbound call with and the call originates from my SIP phone (a Snom 320). The FreeSWITCH logs show that do_2833() in switch_rtp.c isn't called in this case. I'll gladly provide further details if and when anyone has a chance to investigate, assuming that it isn't desired behaviour (which in my opinion it isn't). From pablosaro at gmail.com Fri May 8 22:53:08 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Sat, 9 May 2009 02:53:08 -0300 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <636F7D02-E6E2-419F-9F96-AB2AC1A893F8@jerris.com> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> <636F7D02-E6E2-419F-9F96-AB2AC1A893F8@jerris.com> Message-ID: <247f8100905082253t171aa9b9m285510cc6d2926d4@mail.gmail.com> Jason: IMHO, it is related with the following kernel options: CONFIG_HZ, CONFIG_NO_HZ and CONFIG_HIGH_RES_TIMERS. Take a look at those options in your kernel and try modifying them until get the desired result. Google that options and you will find lots of discussions that will clarify your mind. Here you will find a simple explanation: http://www.smk.co.za/2007/07/21/a-tickless-kernel/ My configuration is 100 Hz + tickless + High-Res timers and everything goes perfect. When system is idle and FS has no calls, I see my CPU at 0%. Hope it helps. Pablo On Thu, Apr 30, 2009 at 3:48 PM, Michael Jerris wrote: > Can you point out any place we do sub milli second sleeps? The timer > thread should be doing 1ms, I can't think of any that would be less. > > MIke > > On Apr 30, 2009, at 2:28 PM, Pawe? Pier?cionek wrote: > > > Hi, > > > > With really old kernels (100Hz) if You do sleep(1ms) You sleep for > > 10ms on average. > > With enterprise kernels (250Hz) Your sleep resolution increases by a > > factor of 4. > > With fresh kernels (1000Hz) You get real 1ms timer resolution - > > 10fold increase compared to old kernels. > > > > With tickless You get whatever resolution You want - eg when You > > sleep for 100 microseconds(micro not mili) then You get exactly what > > You wish for. > > > > Now for reasons I do no try to understand :) there are a lot of > > really short sleeps and fast timers in FreeSwitch - like 100 > > micro(1/10th of a ms). > > So with CentOS such a 100 microsecond sleep cannot "fire" faster > > then 250 times a second. > > With tickless kernel same 100 microsecond sleep "fires" 10k times a > > second. > > > > Pawel, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/7658fa36/attachment.html From jason at jasonjgw.net Fri May 8 23:14:24 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 9 May 2009 16:14:24 +1000 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <247f8100905082253t171aa9b9m285510cc6d2926d4@mail.gmail.com> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> <636F7D02-E6E2-419F-9F96-AB2AC1A893F8@jerris.com> <247f8100905082253t171aa9b9m285510cc6d2926d4@mail.gmail.com> Message-ID: <20090509061424.GA18123@jdc.jasonjgw.net> Pablo Hernan Saro wrote: > IMHO, it is related with the following kernel options: CONFIG_HZ, > CONFIG_NO_HZ and CONFIG_HIGH_RES_TIMERS. > Take a look at those options in your kernel and try modifying them until get > the desired result. Google that options and you will find lots of > discussions that will clarify your mind. Here you will find a simple > explanation: http://www.smk.co.za/2007/07/21/a-tickless-kernel/ Thank you for the references. I think I'll modify my kernel parameters in the grub configuration, since I am using Debian kernels at the moment. (I do know how to compile my own, which I will gladly do if it becomes necessary). From larclap at yahoo.com Sat May 9 07:44:40 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 9 May 2009 07:44:40 -0700 Subject: [Freeswitch-users] Total noob question In-Reply-To: <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> Message-ID: <039201c9d0b4$af230b90$0d6922b0$@com> Thanks, Michael, CentOS worked like a charm. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 08, 2009 2:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Well, Tony likes to say that you can compile FreeSWITCH anywhere but he only supports it if the name of the OS rhymes with Mentos. :) We have had much success with CentOS 5.x installs. I can highly recommend it. -MC On Fri, May 8, 2009 at 1:46 PM, Lars Zeb wrote: Sorry to report that the results are the same. I executed the 'make' command after the 'wget' and captured stderr and stdout to a file. I uploaded it to http://rapidshare.com/files/230726648/openSuse_11.1_Freeswitch-1.0.4pre7_log .zip.html if it's of any use. The mod_cluechoo errors persist. The only mods in the mod folder are mod_amr.(la & so) and mod-cdr_csv.(la & so). Maybe I should swap another OS. I'm willing to try anything to get freeswitch up. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 08, 2009 10:26 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Okay, time for a fresh checkout. If you are on Linux then I recommend that you completely "rm -fr" your freeswitch source directory, then do the quick and dirty install: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Let us know if that works... -MC On Fri, May 8, 2009 at 10:20 AM, Lars Zeb wrote: The mod modules are still not built. I tried with your suggestion. making install mod_cluechoo Compiling mod_cluechoo.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_cluechoo.c -fPIC -DPIC -o .libs/mod_cluechoo.o cc1: warnings being treated as errors mod_cluechoo.c: In function 'go': mod_cluechoo.c:188: error: call to function 'add_D51' without a real prototype mod_cluechoo.c:45: note: 'add_D51' was declared here mod_cluechoo.c:190: error: call to function 'add_sl' without a real prototype mod_cluechoo.c:46: note: 'add_sl' was declared here mod_cluechoo.c: In function 'vgo': mod_cluechoo.c:313: error: call to function 'add_D51' without a real prototype mod_cluechoo.c:45: note: 'add_D51' was declared here mod_cluechoo.c:315: error: call to function 'add_sl' without a real prototype mod_cluechoo.c:46: note: 'add_sl' was declared here mod_cluechoo.c: In function 'add_sl': mod_cluechoo.c:369: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:370: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:370: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:371: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:371: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:373: error: call to function 'add_smoke' without a real prototype mod_cluechoo.c:48: note: 'add_smoke' was declared here mod_cluechoo.c: In function 'add_D51': mod_cluechoo.c:419: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:420: error: call to function 'add_man' without a real prototype mod_cluechoo.c:47: note: 'add_man' was declared here mod_cluechoo.c:422: error: call to function 'add_smoke' without a real prototype mod_cluechoo.c:48: note: 'add_smoke' was declared here make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_cluechoo-install] Error 1 make[2]: *** [install-recursive] Error 1 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + . + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 08, 2009 8:30 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars, May I suggest this process: make clean && svn up && ./bootstrap.sh && ./configure && make install It's one step less intensive than starting with a fresh checkout. I just did that on four systems last night that all had a buggy build with cluechoo and file_string and now all four are humming along nicely. -MC On Fri, May 8, 2009 at 6:54 AM, Lars Zeb wrote: In launching freeswitch, I noticed the errors below, plus many more I did not copy here. Also the mod directory had only two modules in it, so I rebuilt from the tarball again. Same results. I did ./configure, make and make install. What have I done wrong? Thanks, Lars 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_console.so **/usr/local/freeswitch/mod/mod_console.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_logfile.so **/usr/local/freeswitch/mod/mod_logfile.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_enum.so **/usr/local/freeswitch/mod/mod_enum.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:314 load_config() Adding default template. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template sql. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template example. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template snom. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template linksys. 2009-05-08 05:52:32 [DEBUG] mod_cdr_csv.c:359 load_config() Adding template asterisk. 2009-05-08 05:52:32 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_cdr_csv] 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_event_socket.so **/usr/local/freeswitch/mod/mod_event_socket.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_loopback.so **/usr/local/freeswitch/mod/mod_loopback.so: cannot open shared object file: No such file or directory** 2009-05-08 05:52:32 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_commands.so **/usr/local/freeswitch/mod/mod_commands.so: cannot open shared object file: No such file or directory** ... FreeSWITCH Version 1.0.4pre7 (13238M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] fs:/usr/local/freeswitch/mod # l total 104 drwxr-xr-x 2 root root 4096 2009-05-08 06:23 ./ drwxr-xr-x 13 root root 4096 2009-05-08 06:23 ../ -rwxr-xr-x 1 root root 37098 2009-05-08 06:23 mod_amr.so* -rwxr-xr-x 1 root root 52950 2009-05-08 06:23 mod_cdr_csv.so* make[5]: *** [mod_cluechoo.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cluechoo-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Making install in build +-------- FreeSWITCH install Complete ----------+ + FreeSWITCH has been successfully installed. + + + ... + + +-----------------------------------------------+ make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Thursday, May 07, 2009 6:25 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Total noob question Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? Maybe the mod_commands module wasn't loaded. Check your logs, and try "load mod_commands" from the FreeSWITCH console. That module should have been loaded unless there is something seriously wrong with the build or installation process. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/f67ab7e8/attachment-0001.html From larclap at yahoo.com Sat May 9 07:52:44 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 9 May 2009 07:52:44 -0700 Subject: [Freeswitch-users] New install can't register softphone In-Reply-To: <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> Message-ID: <039701c9d0b5$cf961fb0$6ec25f10$@com> I've just installed fs-1.0.4pre7 quick-and-dirty and tried to connect a softphone to fs, but the phone refuses to register. I have tried two different ones (EyeBeam and Zoiper), but neither can register. I have turned off all firewalls(XP firewall, Outpost) on the client machine where the softphone is installed. I can ping the freeswitch machine from the client and I can ping the client machine from the freeswitch machine. The wireshark conversation is: 12 1.539643 192.168.10.11 192.168.10.29 SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29;transport=UDP 13 1.539676 192.168.10.11 192.168.10.29 SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29;transport=UDP 14 1.539722 192.168.10.11 192.168.10.29 SIP Request: REGISTER sip:192.168.10.29;transport=UDP 15 1.539757 192.168.10.11 192.168.10.29 SIP Request: REGISTER sip:192.168.10.29;transport=UDP 16 1.540027 192.168.10.29 192.168.10.11 ICMP Destination unreachable (Host administratively prohibited) where 192.168.10.11 is the client machine and 192.168.10.29 is freeswitch. Freeswitch is running on CentOS 5.3. What is keeping the freeswitch machine from pinging the client? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/387d98b3/attachment.html From anthony.minessale at gmail.com Sat May 9 08:14:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 9 May 2009 10:14:53 -0500 Subject: [Freeswitch-users] New install can't register softphone In-Reply-To: <039701c9d0b5$cf961fb0$6ec25f10$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> <039701c9d0b5$cf961fb0$6ec25f10$@com> Message-ID: <191c3a030905090814n7d052cb2n7b1ea8fa413fd93b@mail.gmail.com> probably iptables try /etc/init.d/iptables stop if it starts working, go back and fix the rules or leave it off if you are already in a private network. On Sat, May 9, 2009 at 9:52 AM, Lars Zeb wrote: > I?ve just installed fs-1.0.4pre7 quick-and-dirty and tried to connect a > softphone to fs, but the phone refuses to register. I have tried two > different ones (EyeBeam and Zoiper), but neither can register. > > > > I have turned off all firewalls(XP firewall, Outpost) on the client machine > where the softphone is installed. I can ping the freeswitch machine from > the client and I can ping the client machine from the freeswitch machine. > > > > The wireshark conversation is: > > > > 12 1.539643 192.168.10.11 192.168.10.29 > SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29 > ;transport=UDP > > 13 1.539676 192.168.10.11 192.168.10.29 > SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29 > ;transport=UDP > > 14 1.539722 192.168.10.11 192.168.10.29 > SIP Request: REGISTER sip:192.168.10.29;transport=UDP > > 15 1.539757 192.168.10.11 192.168.10.29 > SIP Request: REGISTER sip:192.168.10.29;transport=UDP > > 16 1.540027 192.168.10.29 192.168.10.11 > ICMP Destination unreachable (Host administratively prohibited) > > > > where 192.168.10.11 is the client machine and 192.168.10.29 is freeswitch. > Freeswitch is running on CentOS 5.3. > > > > What is keeping the freeswitch machine from pinging the client? > > > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/3241d69a/attachment.html From larclap at yahoo.com Sat May 9 09:48:05 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 9 May 2009 09:48:05 -0700 Subject: [Freeswitch-users] New install can't register softphone In-Reply-To: <191c3a030905090814n7d052cb2n7b1ea8fa413fd93b@mail.gmail.com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> <039701c9d0b5$cf961fb0$6ec25f10$@com> <191c3a030905090814n7d052cb2n7b1ea8fa413fd93b@mail.gmail.com> Message-ID: <000301c9d0c5$ecf98500$c6ec8f00$@com> Logged onto root and typed: /etc/init.d/iptables save /etc/init.d/iptables stop Message: Unloading iptables modules ps -ef shows no iptables running Still cannot register phone From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, May 09, 2009 8:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] New install can't register softphone probably iptables try /etc/init.d/iptables stop if it starts working, go back and fix the rules or leave it off if you are already in a private network. On Sat, May 9, 2009 at 9:52 AM, Lars Zeb wrote: I've just installed fs-1.0.4pre7 quick-and-dirty and tried to connect a softphone to fs, but the phone refuses to register. I have tried two different ones (EyeBeam and Zoiper), but neither can register. I have turned off all firewalls(XP firewall, Outpost) on the client machine where the softphone is installed. I can ping the freeswitch machine from the client and I can ping the client machine from the freeswitch machine. The wireshark conversation is: 12 1.539643 192.168.10.11 192.168.10.29 SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29 ;transport=UDP 13 1.539676 192.168.10.11 192.168.10.29 SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29 ;transport=UDP 14 1.539722 192.168.10.11 192.168.10.29 SIP Request: REGISTER sip:192.168.10.29;transport=UDP 15 1.539757 192.168.10.11 192.168.10.29 SIP Request: REGISTER sip:192.168.10.29;transport=UDP 16 1.540027 192.168.10.29 192.168.10.11 ICMP Destination unreachable (Host administratively prohibited) where 192.168.10.11 is the client machine and 192.168.10.29 is freeswitch. Freeswitch is running on CentOS 5.3. What is keeping the freeswitch machine from pinging the client? Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/a5a70ee7/attachment-0001.html From larclap at yahoo.com Sat May 9 10:48:12 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 9 May 2009 10:48:12 -0700 Subject: [Freeswitch-users] New install can't register softphone In-Reply-To: <000301c9d0c5$ecf98500$c6ec8f00$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> <039701c9d0b5$cf961fb0$6ec25f10$@com> <191c3a030905090814n7d052cb2n7b1ea8fa413fd93b@mail.gmail.com> <000301c9d0c5$ecf98500$c6ec8f00$@com> Message-ID: <001101c9d0ce$5301ff50$f905fdf0$@com> Sorry, I rebooted both machines and now can hear the moh. Thanks. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Saturday, May 09, 2009 9:48 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] New install can't register softphone Logged onto root and typed: /etc/init.d/iptables save /etc/init.d/iptables stop Message: Unloading iptables modules ps -ef shows no iptables running Still cannot register phone From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, May 09, 2009 8:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] New install can't register softphone probably iptables try /etc/init.d/iptables stop if it starts working, go back and fix the rules or leave it off if you are already in a private network. On Sat, May 9, 2009 at 9:52 AM, Lars Zeb wrote: I've just installed fs-1.0.4pre7 quick-and-dirty and tried to connect a softphone to fs, but the phone refuses to register. I have tried two different ones (EyeBeam and Zoiper), but neither can register. I have turned off all firewalls(XP firewall, Outpost) on the client machine where the softphone is installed. I can ping the freeswitch machine from the client and I can ping the client machine from the freeswitch machine. The wireshark conversation is: 12 1.539643 192.168.10.11 192.168.10.29 SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29 ;transport=UDP 13 1.539676 192.168.10.11 192.168.10.29 SIP Request: SUBSCRIBE sip:1000 at 192.168.10.29 ;transport=UDP 14 1.539722 192.168.10.11 192.168.10.29 SIP Request: REGISTER sip:192.168.10.29;transport=UDP 15 1.539757 192.168.10.11 192.168.10.29 SIP Request: REGISTER sip:192.168.10.29;transport=UDP 16 1.540027 192.168.10.29 192.168.10.11 ICMP Destination unreachable (Host administratively prohibited) where 192.168.10.11 is the client machine and 192.168.10.29 is freeswitch. Freeswitch is running on CentOS 5.3. What is keeping the freeswitch machine from pinging the client? Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/1a45c72b/attachment.html From larclap at yahoo.com Sat May 9 17:53:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 9 May 2009 17:53:03 -0700 Subject: [Freeswitch-users] Asterisk and Freeswitch co-existence? In-Reply-To: <001101c9d0ce$5301ff50$f905fdf0$@com> References: <23426138.post@talk.nabble.com> <01e601c9cf7a$431d8e60$c958ab20$@com> <20090508012515.GA18966@jdc.jasonjgw.net> <026d01c9cfe4$89f41e20$9ddc5a60$@com> <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> <039701c9d0b5$cf961fb0$6ec25f10$@com> <191c3a030905090814n7d052cb2n7b1ea8fa413fd93b@mail.gmail.com> <000301c9d0c5$ecf98500$c6ec8f00$@com> <001101c9d0ce$5301ff50$f905fdf0$@com> Message-ID: <003401c9d109$ac8c5a80$05a50f80$@com> I am new to Freeswitch. Before starting with Freeswitch, I had, and still do, an Asterisk (Switchvox) on the LAN. What changes to the Freeswitch config files do I need to make so that it can co-exist with the Asterisk box on the same LAN? I don't mean collaborate or interoperate, just co-exist. I know I need to change the SIP port number. Is there a document which describes this environment? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090509/c7fa14fc/attachment.html From jason at jasonjgw.net Sat May 9 18:14:11 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 10 May 2009 11:14:11 +1000 Subject: [Freeswitch-users] Install without example configurations In-Reply-To: <5DA33B25-34EB-4DBC-8AF7-0546012749B5@freeswitch.org> References: <5d3e0dc60905081322r2534f0e9ub340c4792db7f6e6@mail.gmail.com> <5DA33B25-34EB-4DBC-8AF7-0546012749B5@freeswitch.org> Message-ID: <20090510011411.GA15109@jdc.jasonjgw.net> Brian West wrote: > > but You'll have to do it all by hand its better off to strip down the > config once you understand how FreeSWITCH works. Correct. The introduction on the wiki gives a helpful overview of the structure of the default configuration files - start with freeswitch.xml and follow the inclusions. Since the default configuration was similar in many respect to what I needed, it was quicker to modify it than to write a completely new set of configuration files. However, your needs may be different, and there are circumstances in which writing your own files from scratch would be the best solution. From jason at jasonjgw.net Sat May 9 18:16:45 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 10 May 2009 11:16:45 +1000 Subject: [Freeswitch-users] Asterisk and Freeswitch co-existence? In-Reply-To: <003401c9d109$ac8c5a80$05a50f80$@com> References: <87f2f3b90905080829mca8c072r20f2857bf6596ee2@mail.gmail.com> <02b901c9d001$5622b670$02682350$@com> <87f2f3b90905081026l510b7a2dx279cbd9f452867ee@mail.gmail.com> <02ee01c9d01e$148fe260$3dafa720$@com> <87f2f3b90905081400p5dccec2ble25e84d9a682683c@mail.gmail.com> <039701c9d0b5$cf961fb0$6ec25f10$@com> <191c3a030905090814n7d052cb2n7b1ea8fa413fd93b@mail.gmail.com> <000301c9d0c5$ecf98500$c6ec8f00$@com> <001101c9d0ce$5301ff50$f905fdf0$@com> <003401c9d109$ac8c5a80$05a50f80$@com> Message-ID: <20090510011645.GB15109@jdc.jasonjgw.net> Lars Zeb wrote: > What changes to the Freeswitch config files do I need to make so that it can > co-exist with the Asterisk box on the same LAN? I don't mean collaborate or > interoperate, just co-exist. I know I need to change the SIP port number. Why do you need to change the SIP port? If Asterisk and FreeSITCH are running on different machines, you shouldn't have to modify anything, other than to configure phones for use with FreeSWITCH. If they're running on the same machine, then you will need to change the SIP port of one or the other, and that's all you should need to change. From pepus at atlas.cz Sat May 9 18:19:58 2009 From: pepus at atlas.cz (Josef) Date: Sun, 10 May 2009 03:19:58 +0200 Subject: [Freeswitch-users] managed contact list Message-ID: <4A062BBE.9080407@atlas.cz> Hello, does freeswitch somehow support server-managed contact list? I'm unable to force all users to manually add each other in their lists and would welcome if this could be managed from the server... Thanks, Josef From adam.falcone at gmail.com Sun May 10 10:20:32 2009 From: adam.falcone at gmail.com (Adam Falcone) Date: Sun, 10 May 2009 10:20:32 -0700 Subject: [Freeswitch-users] Running FreeSwitch in the background Message-ID: <8ECA0A5A-E37A-4B45-ADE7-81060CC63119@gmail.com> after starting freeswitch I am able to receive incoming calls but if I let freeswitch run in the background it stops picking up incoming calls. The only way I am able to get freeswitch to recognize incoming calls again is by making an outbound call and it then afterwards receives inbound calls. Not sure what if any log information would be beneficial to post here. I am new to freeswitch. From krice at suspicious.org Sun May 10 11:02:56 2009 From: krice at suspicious.org (Ken Rice) Date: Sun, 10 May 2009 13:02:56 -0500 Subject: [Freeswitch-users] Running FreeSwitch in the background In-Reply-To: <8ECA0A5A-E37A-4B45-ADE7-81060CC63119@gmail.com> Message-ID: How are you running it in the background? Freeswitch -nc starts it in the background and it should run fine there > From: Adam Falcone > Reply-To: > Date: Sun, 10 May 2009 10:20:32 -0700 > To: > Subject: [Freeswitch-users] Running FreeSwitch in the background > > after starting freeswitch I am able to receive incoming calls but if I > let freeswitch run in the background it stops picking up incoming > calls. The only way I am able to get freeswitch to recognize incoming > calls again is by making an outbound call and it then afterwards > receives inbound calls. Not sure what if any log information would be > beneficial to post here. I am new to freeswitch. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From diego.viola at gmail.com Sun May 10 12:06:58 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 10 May 2009 15:06:58 -0400 Subject: [Freeswitch-users] mod_nibblebill question In-Reply-To: <86a32abc0905081208u76345237q190559b8f0b161d4@mail.gmail.com> References: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> <86a32abc0905081208u76345237q190559b8f0b161d4@mail.gmail.com> Message-ID: <86a32abc0905101206j76c10a21oe059b360118e5254@mail.gmail.com> Darren Schreiber to me That won't work. The code isn't written yet. Give mea week or two to finish that. On Fri, May 8, 2009 at 3:08 PM, Diego Viola wrote: > I have set these actions: > > > > > But when it reaches 0 cash it doesn't hangup :(. > > On Fri, May 8, 2009 at 2:34 PM, Diego Viola wrote: >> Oh I see that it has it already :D >> >> >> >> >> - >> >> >> >> >> Diego >> >> On Fri, May 8, 2009 at 1:20 PM, Diego Viola wrote: >>> Hi everyone, >>> >>> I'm currently developing a calling card application that uses event >>> socket and mod_nibblebill to bill calls. Well, the question is: can >>> mod_nibblebill disconnect a call when the balance is depleted, or when >>> it reaches 0 cash? >>> >>> The wiki says: >>> >>> "Allow for disconnecting or re-routing calls when balance is depleted." >>> >>> But then at the bottom in "future goals" it says: >>> >>> "We don't yet warn or terminate calls when they've reached their limits." >>> >>> So I'm confused, I also don't see any options in how to set that. In >>> case that mod_nibblebill doesn't support this yet, how hard would it >>> be to add this? I'm willing to do it, I guess it's a variable on the >>> XML config and then read that from the mod_nibblebill.c, and do some >>> logic there. >>> >>> Just wondering if anyone had some experience with this, and if the >>> feature is already there. >>> >>> Thanks, >>> >>> Diego >>> >> > From electromech at electromech.info Sun May 10 17:12:36 2009 From: electromech at electromech.info (Nilesh J. Vaghela) Date: Mon, 11 May 2009 05:42:36 +0530 Subject: [Freeswitch-users] basic setup and test Message-ID: <4A076D74.40301@electromech.info> Hi, I am new to voip and freeswitch too. I am trying to configure freeswitch initially for testing purpose. I configured freeswitch on Fedora10 working ok. With default configuration I can use EKIGA and X-Lite. I want is configure freeswitch as soft EPABX. Is there any small or basic howto any body can suggest. -- Nilesh Vaghela ElectroMech Redhat Channel Partner and Training Partner 16, Sun Rise complex, Nr. Mansi cross Road, Satellite Rd, Ahmedabad 25, The Emperor, Fatehgunj, Baroda. www.electromech.info From dftoro at yahoo.com Sun May 10 18:17:09 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sun, 10 May 2009 18:17:09 -0700 (PDT) Subject: [Freeswitch-users] get call durantion Message-ID: <663778.57624.qm@web33503.mail.mud.yahoo.com> Hi, How can I get call durantion after bridge application ? ?I tried with billsec and duration but I don't get any value. ? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090510/7ae7b4f5/attachment.html From msc at freeswitch.org Sun May 10 21:27:49 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 10 May 2009 21:27:49 -0700 Subject: [Freeswitch-users] get call durantion In-Reply-To: <663778.57624.qm@web33503.mail.mud.yahoo.com> References: <663778.57624.qm@web33503.mail.mud.yahoo.com> Message-ID: <065FEF43-800F-4571-B2FC-D9F147E0F078@freeswitch.org> Do you mean from the CDR? I recommend XML CDRs because they give tons of information. If you are talking about gathering this stuff midcall then you'll need to supply more information about your setup. -MC Sent from my iPhone On May 10, 2009, at 6:17 PM, Diego Toro wrote: > Hi, > How can I get call durantion after bridge application ? I tried > with billsec and duration but I don't get any value. > > Thank you > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090510/6471acf8/attachment.html From msc at freeswitch.org Sun May 10 21:36:54 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 10 May 2009 21:36:54 -0700 Subject: [Freeswitch-users] basic setup and test In-Reply-To: <4A076D74.40301@electromech.info> References: <4A076D74.40301@electromech.info> Message-ID: <274D91E7-28FD-4D85-B4C5-20D6C580B746@freeswitch.org> If you installed the default configuration then you've already got a PBX. I recommend looking at the default.xml file in conf/dialplan and also the user setup files like 1000.xml in conf/directory/default. The default config has 20 users already setup including voicemail. It also has examples on how to setup a service provider (gateway) for VoIP. See conf/sip_profiles for more information. Also, read up on this topic on the wiki. Sip profiles are very confusing at first. The best thing you can do is set up a system with the default config and play with it. You'll soon learn lots of things that you can do. -MC Sent from my iPhone On May 10, 2009, at 5:12 PM, "Nilesh J. Vaghela" wrote: > Hi, > > I am new to voip and freeswitch too. > > I am trying to configure freeswitch initially for testing purpose. > > I configured freeswitch on Fedora10 working ok. > > With default configuration I can use EKIGA and X-Lite. > > I want is configure freeswitch as soft EPABX. > > Is there any small or basic howto any body can suggest. > > > -- > Nilesh Vaghela > ElectroMech > Redhat Channel Partner and Training Partner > 16, Sun Rise complex, > Nr. Mansi cross Road, > Satellite Rd, Ahmedabad > 25, The Emperor, Fatehgunj, Baroda. > www.electromech.info > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Richard.Lamkin at mettoni.com Mon May 11 01:54:53 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Mon, 11 May 2009 09:54:53 +0100 Subject: [Freeswitch-users] Trunk @13273 - build on VS2008 fails - since switch_xml.h was checked in In-Reply-To: References: Message-ID: <3181A30B8C35AB4AA8577B78DDF46138050141F9@nickel.mettonigroup.com> Build on fails @ trunk =13273 My last working build @ trunk = 13231 Looks like the addition of the directives _Out_, _In_, _In_opt_, .... has caused the problem. Unfortunately I'm not familiar that with the code or these directives to suggest a solution, apart from the obvious - roll back. Regards Richard Lamkin Richard.lamkin at mettonigroup.com Error 787 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 mod_managed Error 816 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 mod_managed Error 846 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 mod_managed Error 1372 error C2220: warning treated as error - no 'object' file generated d:\wip\freeswitch\trunk\src\include\switch_xml.h 107 FreeSwitchConsole 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6510: Invalid annotation: 'NullTerminated' property may only be used on buffers whose elements are of integral or pointer type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : error C2220: warning treated as error - no 'object' file generated 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6510: Invalid annotation: 'NullTerminated' property may only be used on buffers whose elements are of integral or pointer type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(354) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(347) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(354) : warning C6504: Invalid annotation: 'Null' property may only be used on values of pointer, pointer-to-member, array, or reference type 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(347) : warning C6516: Invalid annotation: no properties specified for PreAttribute attribute ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* From mike at yes.net.ua Mon May 11 02:17:44 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Mon, 11 May 2009 12:17:44 +0300 Subject: [Freeswitch-users] Stops accepting calls when idle for four minutes In-Reply-To: <461764FC-2E50-44F9-ABD4-D4CEBD32FB35@cgicommunications.com> References: <90210A67-07BD-40E4-AD49-6D195C7FDC67@cgicommunications.com> <666386713.20090508194901@yes.net.ua> <461764FC-2E50-44F9-ABD4-D4CEBD32FB35@cgicommunications.com> Message-ID: <10812511.20090511121744@yes.net.ua> Hello Greg, Should be no problem - only few more bytes of bandwidth used. Friday, May 8, 2009 9:23:49 PM, you wrote: GT> That solved it! Is there any downside to this method of keeping the nat binding alive? GT> -- GT> Greg GT> On May 8, 2009, at 12:49 PM, Mike Tkachuk wrote: GT> Hello Greg, GT> It's a NAT box issue. Nat bindings expire if no activity. GT> Try adding a: GT> GT> to your gateway params. GT> But to be honest it's flowroute duty to keep a connection alive by GT> sending keepalives. GT> -- GT> Mike Tkachuk -- Mike Tkachuk From helmut.kuper at ewetel.de Mon May 11 02:42:39 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 11 May 2009 11:42:39 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <5800526b0905080831n31c745d6r948536633a033960@mail.gmail.com> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> <4A03FB80.2070001@ewetel.de> <4A041F01.1010308@ewetel.de> <4A043435.30800@ewetel.de> <5800526b0905080718w214d8933rdbb3febea12dd6e4@mail.gmail.com> <5800526b0905080831n31c745d6r948536633a033960@mail.gmail.com> Message-ID: <4A07F30F.1050408@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Carlos, On 08.05.2009 17:31, Carlos Talbot wrote: > Nevermind. I figured it out. It's available as beta on the dms server. indeed, it's a beta FW and quite slow compared to FW7.X.X. Look and Feel of internal webpage as well of some keys (like '?') is same as the Snom 820/870 series. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKB/MP4tZeNddg3dwRAlK3AJ46ZxZ/7fhUssbeTjFs4oi7FaKH2ACfQLZB uZ/Wfb1m1Zy7x9w5gM4Yw34= =6lYB -----END PGP SIGNATURE----- From kjv at ken-ton.com Mon May 11 06:02:28 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Mon, 11 May 2009 09:02:28 -0400 Subject: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel In-Reply-To: <20090509061424.GA18123@jdc.jasonjgw.net> References: <20090427010053.GA20422@jdc.jasonjgw.net> <2C723DE5-FEC3-478F-9B4E-F36AA5092E4F@voiceworks.pl> <87fxfqk87f.fsf@rimspace.net> <12F37DAF-B03B-4548-8630-F844FDE5A821@voiceworks.pl> <636F7D02-E6E2-419F-9F96-AB2AC1A893F8@jerris.com> <247f8100905082253t171aa9b9m285510cc6d2926d4@mail.gmail.com> <20090509061424.GA18123@jdc.jasonjgw.net> Message-ID: <99AE2B05-E1C8-4361-BB3D-D6C262BAC70E@ken-ton.com> Folks; Bear in mind that the frequency is (X)Hz * (num cores), hence saying 100Hz on a dual core winds up being 200Hz. My setup is 250Hz on a Dual-Core and the quality is perfect. Oh, btw folks, don't attempt to do anything involving QOS (be it TBF, CBQ, HTB, or whatnot) on anything less than kernel 2.6.28.4 I don't know why exactly that is, but extensive testing here in the lab showed that this was entirely FUBAR until 2.6.25.7 where it got better, but not perfect until 2.6.28.4 (I may not be exact on the revisions, but close enough...) There's also some options in the kernel that you must disable (not compile in) if you expect packet shaping to work. If there's interest in this, e-mail me directly and I'll see if I can toss it into the FS Wiki some time this upcoming weekend. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On May 9, 2009, at 2:14 AM, Jason White wrote: > Pablo Hernan Saro wrote: > >> IMHO, it is related with the following kernel options: CONFIG_HZ, >> CONFIG_NO_HZ and CONFIG_HIGH_RES_TIMERS. >> Take a look at those options in your kernel and try modifying them >> until get >> the desired result. Google that options and you will find lots of >> discussions that will clarify your mind. Here you will find a simple >> explanation: http://www.smk.co.za/2007/07/21/a-tickless-kernel/ > > Thank you for the references. I think I'll modify my kernel > parameters in the > grub configuration, since I am using Debian kernels at the moment. > (I do know > how to compile my own, which I will gladly do if it becomes > necessary). > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/3adf0b98/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/3adf0b98/attachment.bin From anthony.minessale at gmail.com Mon May 11 06:07:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 May 2009 08:07:25 -0500 Subject: [Freeswitch-users] Trunk @13273 - build on VS2008 fails - since switch_xml.h was checked in In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138050141F9@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF46138050141F9@nickel.mettonigroup.com> Message-ID: <191c3a030905110607o484c1dbbj9b0a1f39febfb712@mail.gmail.com> Issues of this type must be reported to jira http://jira.freeswitch.org On Mon, May 11, 2009 at 3:54 AM, Richard Lamkin wrote: > Build on fails @ trunk =13273 > My last working build @ trunk = 13231 > > Looks like the addition of the directives _Out_, _In_, _In_opt_, .... > has caused the problem. Unfortunately I'm not familiar that with the > code or these directives to suggest a solution, apart from the obvious - > roll back. > > Regards > Richard Lamkin > Richard.lamkin at mettonigroup.com > > Error 787 error C2220: warning treated as error - no 'object' file > generated d:\wip\freeswitch\trunk\src\include\switch_xml.h > 107 mod_managed > Error 816 error C2220: warning treated as error - no 'object' file > generated d:\wip\freeswitch\trunk\src\include\switch_xml.h > 107 mod_managed > Error 846 error C2220: warning treated as error - no 'object' file > generated d:\wip\freeswitch\trunk\src\include\switch_xml.h > 107 mod_managed > Error 1372 error C2220: warning treated as error - no 'object' file > generated d:\wip\freeswitch\trunk\src\include\switch_xml.h > 107 FreeSwitchConsole > > > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning > C6510: Invalid annotation: 'NullTerminated' property may only be used on > buffers whose elements are of integral or pointer type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : error C2220: > warning treated as error - no 'object' file generated > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(107) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning > C6510: Invalid annotation: 'NullTerminated' property may only be used on > buffers whose elements are of integral or pointer type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(173) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(212) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(354) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 82>d:\wip\freeswitch\trunk\src\include\switch_xml.h(347) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(221) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(256) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(309) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(322) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(354) : warning > C6504: Invalid annotation: 'Null' property may only be used on values of > pointer, pointer-to-member, array, or reference type > 83>d:\wip\freeswitch\trunk\src\include\switch_xml.h(347) : warning > C6516: Invalid annotation: no properties specified for PreAttribute > attribute > > > > > > ************************************************************************* > This email and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you have received this email in error please notify > the system manager. http://www.mettoni.com > > Mettoni Ltd > Registered in England and Wales: 4485956 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/50f6baf3/attachment-0001.html From peter.olsson at visionutveckling.se Mon May 11 06:34:05 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 15:34:05 +0200 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> I'm implementing an IVR solution for FreeSWITCH, but I have a little problem with audio playback. I'm just calling into application park, and then handle the flow using the event socket. All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after eachother. Between each file I can sometimes here a little "click", even though I'm 100% sure that this is not from my files. My guess is that it might be caused of the fact that no RTP is sent at all when the phrase is not playing. If I merge the files together before playing them it sounds just fine. Is it possible to make FreeSWITCH send silence frames, even when not needed? I know this is a waste of resources, but it will still make the solution sound much better. Regards Peter Olsson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/7447f76f/attachment.html From peter.olsson at visionutveckling.se Mon May 11 06:39:11 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 15:39:11 +0200 Subject: [Freeswitch-users] Audi record using uuid_record Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it's just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call "uuid_record UUID start c:\test.PCMA". According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/d955c88b/attachment.html From brian at freeswitch.org Mon May 11 06:57:56 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 08:57:56 -0500 Subject: [Freeswitch-users] Audi record using uuid_record In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> Message-ID: <96B8A1F8-FED8-4C77-A63D-1CF6E7E175B3@freeswitch.org> Chances are if both legs are NOT already alaw you'll need to record it with .wav or .al files. .PCMU or .PCMA are native file formats if any transcoding is taking place you probably can't get way with .PCMA. /b On May 11, 2009, at 8:39 AM, Peter Olsson wrote: > Hello again, > > I also have a problem when I try to record messages. I record > to .PCMA-files, and the file is created perfectly. But it?s just > distorted audio in it. It sounds to me that there might be a codec > issue. The media stream is PCMA all the way from the phone to > FreeSWITCH, and to start recording I simply call ?uuid_record UUID > start c:\test.PCMA?. > > According to the docs the file should automatically be recorded as > PCMA when the file is named .PCMA. > > Any ideas what I can be doing wrong? > > Regards, > > Peter > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/02376f4f/attachment.html From helmut.kuper at ewetel.de Mon May 11 07:02:47 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 11 May 2009 16:02:47 +0200 Subject: [Freeswitch-users] SRTP Error "auth check failed" In-Reply-To: <4A07F30F.1050408@ewetel.de> References: <4A02F883.9090507@ewetel.de> <75B9EACC-3022-4D67-8E1C-723093ECCD6A@freeswitch.org> <4A03FB80.2070001@ewetel.de> <4A041F01.1010308@ewetel.de> <4A043435.30800@ewetel.de> <5800526b0905080718w214d8933rdbb3febea12dd6e4@mail.gmail.com> <5800526b0905080831n31c745d6r948536633a033960@mail.gmail.com> <4A07F30F.1050408@ewetel.de> Message-ID: <4A083007.5080304@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I can reproduce the error with Snom 370 and FW 7.3.11. An update to 7.3.20 (beta) seems to solve the problem - at least I can't reproduce it with the metioned scenario anymore. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCDAH4tZeNddg3dwRAs29AKCxFN/193R1D+svIjB+Knzzbuhr1wCfQ4rq hMVg7svm2EoaOOexJS5HCMY= =jhJh -----END PGP SIGNATURE----- From Prometheus001 at gmx.net Mon May 11 07:04:53 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 11 May 2009 16:04:53 +0200 Subject: [Freeswitch-users] Audi record using uuid_record In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> Message-ID: <4A083085.3090907@gmx.net> I record them to file.wav and they play perfectly. I think it's recorded in a raw-format here. See: http://www.nabble.com/Recording-ULAW-files-td21587474.html Best regards Peter Peter Olsson schrieb: > > Hello again, > > > > I also have a problem when I try to record messages. I record to > .PCMA-files, and the file is created perfectly. But it?s just > distorted audio in it. It sounds to me that there might be a codec > issue. The media stream is PCMA all the way from the phone to > FreeSWITCH, and to start recording I simply call ?uuid_record UUID > start c:\test.PCMA?. > > > > According to the docs the file should automatically be recorded as > PCMA when the file is named .PCMA. > > > > Any ideas what I can be doing wrong? > > > > Regards, > > > > Peter > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon May 11 07:08:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 09:08:44 -0500 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> Message-ID: <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> Have you tried wav files? /b On May 11, 2009, at 8:34 AM, Peter Olsson wrote: > I?m implementing an IVR solution for FreeSWITCH, but I have a little > problem with audio playback. I?m just calling into application park, > and then handle the flow using the event socket. > > All my audio phrases are .PCMA (8KHz a-law), and I play lots of > files after eachother. Between each file I can sometimes here a > little ?click?, even though I?m 100% sure that this is not from my > files. My guess is that it might be caused of the fact that no RTP > is sent at all when the phrase is not playing. If I merge the files > together before playing them it sounds just fine. > > Is it possible to make FreeSWITCH send silence frames, even when not > needed? I know this is a waste of resources, but it will still make > the solution sound much better. > > Regards > > Peter Olsson Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/e391b134/attachment.html From peter.olsson at visionutveckling.se Mon May 11 07:15:54 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 16:15:54 +0200 Subject: [Freeswitch-users] Audi record using uuid_record In-Reply-To: <96B8A1F8-FED8-4C77-A63D-1CF6E7E175B3@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> <96B8A1F8-FED8-4C77-A63D-1CF6E7E175B3@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363831@cooper> In this case it is pure PCMA, all the way from the phone. I just dial in to number 2100 (using SIP, codec PCMA), and then I have a event socket connected that sees the ivr-test flag. I then play some files (PCMA), and then start a recording. I understand that it works with wav, however, the application I'm working on already exists, and it makes lots of trouble for me to change the file format during the recording - I have lots of other parts of the software that needs to be changed as well. I've used yate for this application before this - so everything does exist, I'm just porting it to FreeSWITCH. //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 11 maj 2009 15:58 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Audi record using uuid_record Chances are if both legs are NOT already alaw you'll need to record it with .wav or .al files. .PCMU or .PCMA are native file formats if any transcoding is taking place you probably can't get way with .PCMA. /b On May 11, 2009, at 8:39 AM, Peter Olsson wrote: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it's just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call "uuid_record UUID start c:\test.PCMA". According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a0830b432933740610192! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/abeb79be/attachment.html From peter.olsson at visionutveckling.se Mon May 11 07:20:55 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 16:20:55 +0200 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> Brian, Thanks for the response. No, I didn't try wav files - and I'd prefer to keep the current codec if that's possible. But I could give it a try and see what happens. Do you think it might only be related to the native files in FS? //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 11 maj 2009 16:09 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files Have you tried wav files? /b On May 11, 2009, at 8:34 AM, Peter Olsson wrote: I'm implementing an IVR solution for FreeSWITCH, but I have a little problem with audio playback. I'm just calling into application park, and then handle the flow using the event socket. All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after eachother. Between each file I can sometimes here a little "click", even though I'm 100% sure that this is not from my files. My guess is that it might be caused of the fact that no RTP is sent at all when the phrase is not playing. If I merge the files together before playing them it sounds just fine. Is it possible to make FreeSWITCH send silence frames, even when not needed? I know this is a waste of resources, but it will still make the solution sound much better. Regards Peter Olsson Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a08332632931845617996! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/8b6e8e4f/attachment-0001.html From brian at freeswitch.org Mon May 11 07:24:05 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 09:24:05 -0500 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> Message-ID: <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> I'm not sure. Can you provide me a test file and a known case that you can produce this issue with? /b On May 11, 2009, at 9:20 AM, Peter Olsson wrote: > Brian, > > Thanks for the response. No, I didn?t try wav files ? and I?d prefer > to keep the current codec if that?s possible. But I could give it a > try and see what happens. > > Do you think it might only be related to the native files in FS? > > //Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/9cf0700c/attachment.html From peter.olsson at visionutveckling.se Mon May 11 07:40:34 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 16:40:34 +0200 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4 files, and it sounds great, sometimes it "clicks" louder, somtimes not so much. I could get a Wireshark dump for you, could that help? Regards, Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 11 maj 2009 16:24 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files I'm not sure. Can you provide me a test file and a known case that you can produce this issue with? /b On May 11, 2009, at 9:20 AM, Peter Olsson wrote: Brian, Thanks for the response. No, I didn't try wav files - and I'd prefer to keep the current codec if that's possible. But I could give it a try and see what happens. Do you think it might only be related to the native files in FS? //Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a08362532934700715407! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/7d08cba3/attachment.html From peter.olsson at visionutveckling.se Mon May 11 07:42:43 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 16:42:43 +0200 Subject: [Freeswitch-users] Audi record using uuid_record In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB61D1FC1@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636380D@cooper> <96B8A1F8-FED8-4C77-A63D-1CF6E7E175B3@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB61D1FC1@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363840@cooper> I've confirmed that recording to wav files works just fine. So it seems to be somthing strange with the native files in FS? /Peter Fr?n: Peter Olsson Skickat: den 11 maj 2009 16:16 Till: 'freeswitch-users at lists.freeswitch.org' ?mne: RE: [Freeswitch-users] Audi record using uuid_record In this case it is pure PCMA, all the way from the phone. I just dial in to number 2100 (using SIP, codec PCMA), and then I have a event socket connected that sees the ivr-test flag. I then play some files (PCMA), and then start a recording. I understand that it works with wav, however, the application I'm working on already exists, and it makes lots of trouble for me to change the file format during the recording - I have lots of other parts of the software that needs to be changed as well. I've used yate for this application before this - so everything does exist, I'm just porting it to FreeSWITCH. //Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 11 maj 2009 15:58 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Audi record using uuid_record Chances are if both legs are NOT already alaw you'll need to record it with .wav or .al files. .PCMU or .PCMA are native file formats if any transcoding is taking place you probably can't get way with .PCMA. /b On May 11, 2009, at 8:39 AM, Peter Olsson wrote: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it's just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call "uuid_record UUID start c:\test.PCMA". According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a0830b432933740610192! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/1245f637/attachment-0001.html From brian at freeswitch.org Mon May 11 07:43:03 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 09:43:03 -0500 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> Message-ID: <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> What phone are you using? /b On May 11, 2009, at 9:40 AM, Peter Olsson wrote: > The problem is that it?s not 100% reproducable. Sometimes I can play > 2-3-4 files, and it sounds great, sometimes it ?clicks? louder, > somtimes not so much. I could get a Wireshark dump for you, could > that help? > > Regards, > > Peter > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/d01794fd/attachment.html From peter.olsson at visionutveckling.se Mon May 11 07:51:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 11 May 2009 16:51:28 +0200 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> In this case I'm using Avaya Phones connected to a Avaya CM PBX, which talks SIP to FreeSWITCH. But I'll try to connect a SIP phone directly as well - to see if it makes any difference. I Have a Polycom IP550 I can use for some testing. I've used the exact same setup using yate, and that worked fine, but I think one difference is that it sends rtp even when not playing files, so that's why I thought that could be an issue. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 11 maj 2009 16:43 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files What phone are you using? /b On May 11, 2009, at 9:40 AM, Peter Olsson wrote: The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4 files, and it sounds great, sometimes it "clicks" louder, somtimes not so much. I could get a Wireshark dump for you, could that help? Regards, Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a083a8d32932061814324! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/c92da0fd/attachment.html From msc at freeswitch.org Mon May 11 10:34:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 11 May 2009 10:34:20 -0700 Subject: [Freeswitch-users] Cluecon 2009 News Message-ID: <87f2f3b90905111034g71497662m9b2e623862d2862d@mail.gmail.com> FYI, for those of you keeping up on ClueCon 2009 please visit the latest blog entry: http://cluecon.com/node/29 Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/2b4d3133/attachment.html From larclap at yahoo.com Mon May 11 14:38:16 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 11 May 2009 14:38:16 -0700 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> Message-ID: <013701c9d280$cbdf84c0$639e8e40$@com> I am having difficulty making an outbound call. I have read the wiki many times, but I am missing something. First, after setting up a gateway xml file, should that gateway show in the FS console after issuing the 'reloadxml' command? It does not. Can anyone give me a push? Thanks, Lars API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at 64.105.128.82:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG internal profile sip:mod_sofia at 192.168.10.29:5090 RUNNING (0) 192.168.10.29 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5090 RUNNING (0) default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED in conf/directory/default/flowroute.com.xml: in conf/dialplan/default/02_long_distance.xml: Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Regex (PASS) [Long Distance - flowroute] destination_number(3235551212) =~ /^(1{0,1}\d{10})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action set(effective_caller_id_number=10000000000) Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action bridge(sofia/gateway/flowroute/flowrouteAccount#3235551212) 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1000 at 192.168.10.29:5090) State Change CS_ROUTING -> CS_EXECUTE 2009-05-11 14:05:41 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at 192.168.10.29:5090 [BREAK] 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) State ROUTING going to sleep 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) Running State Change CS_EXECUTE 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) State EXECUTE 2009-05-11 14:05:41 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1000 at 192.168.10.29:5090 SOFIA EXECUTE 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1000 at 192.168.10.29:5090 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(open=true) 2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(use_profile=nat) 2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [use_profile]=[nat] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set_user(default at 192.168.10.29) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/192.168.10.29-spymap/1000/7c7baffc-3e6f-11de-9b6b-7ba002f89c82) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/192.168.10.29-last_dial/1000/3235551212) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/192.168.10.29-last_dial/global/7c7baffc-3e6f-11de-9b6b-7ba002f89 c82) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(effective_caller_id_number=10000000000) 2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [effective_caller_id_number]=[10000000000] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(effective_caller_id_name=FreeSWITCH) 2009-05-11 14:05:41 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [effective_caller_id_name]=[FreeSWITCH] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 bridge(sofia/gateway/flowroute/flowrouteAccount#3235551212) 2009-05-11 14:05:41 [ERR] mod_sofia.c:2531 sofia_outgoing_channel() Invalid Gateway 2009-05-11 14:05:41 [NOTICE] mod_sofia.c:2744 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (N/A) State DESTROY 2009-05-11 14:05:41 [DEBUG] mod_sofia.c:240 sofia_on_destroy() N/A SOFIA DESTROY 2009-05-11 14:05:41 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (N/A) State DESTROY going to sleep 2009-05-11 14:05:41 [ERR] switch_ivr_originate.c:1494 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/492b11bb/attachment-0001.html From brian at freeswitch.org Mon May 11 14:57:37 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 16:57:37 -0500 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <013701c9d280$cbdf84c0$639e8e40$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> Message-ID: <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> Try this ^1?(\d{10})$ and this sofia/gateway/flowroute/flowrouteAccount#1$1 /b On May 11, 2009, at 4:38 PM, Lars Zeb wrote: > in conf/dialplan/default/02_long_distance.xml: > > > > data="effective_caller_id_number=10000000000"/> > > > > > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From mcampbellsmith at gmail.com Mon May 11 15:11:00 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 12 May 2009 08:11:00 +1000 Subject: [Freeswitch-users] How to enable debug in dingaling? Message-ID: <33c87fa30905111511jcf31524la4aa555f3cf9aaf@mail.gmail.com> Hi! I need to enable debug mode in dingaling as I can't see that freeswitch is coming online in gtalk. I have changed the following: changed the loglevel to debug in console.conf.xml changed the debug level to 1 in dingaling.conf to 1 I do not see any xmpp logs in the console or in freeswitch.log file. All I can see in the window is: 2009-05-12 17:56:35 [DEBUG] mod_dingaling.c:1854 init_profile() Started Thread for myfreeswitchname at gmail.com/gtalk at XML 2009-05-12 17:56:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dingaling] 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'dingaling' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_debug' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_pres' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_logout' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_login' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'jingle' Where is the XMPP traces? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/7c317b53/attachment.html From brian at freeswitch.org Mon May 11 15:16:48 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 17:16:48 -0500 Subject: [Freeswitch-users] How to enable debug in dingaling? In-Reply-To: <33c87fa30905111511jcf31524la4aa555f3cf9aaf@mail.gmail.com> References: <33c87fa30905111511jcf31524la4aa555f3cf9aaf@mail.gmail.com> Message-ID: Try "dl_debug on" at the CLI /b On May 11, 2009, at 5:11 PM, Mark Campbell-Smith wrote: > Where is the XMPP traces? > > Thanks Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/48f99463/attachment.html From larclap at yahoo.com Mon May 11 15:23:48 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 11 May 2009 15:23:48 -0700 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> Message-ID: <014e01c9d287$27d5a2e0$7780e8a0$@com> I'm sorry, but do not understand what it is that I should try. Are you saying to change the data attribute in the action command of the dialplan? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, May 11, 2009 2:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't configure outbound call Try this ^1?(\d{10})$ and this sofia/gateway/flowroute/flowrouteAccount#1$1 /b On May 11, 2009, at 4:38 PM, Lars Zeb wrote: > in conf/dialplan/default/02_long_distance.xml: > > > > data="effective_caller_id_number=10000000000"/> > > > > > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon May 11 15:28:00 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 17:28:00 -0500 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <014e01c9d287$27d5a2e0$7780e8a0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> Message-ID: <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> You want to collect 11 or 10 digits and send it out flowroute which requires 1+ number. The 1? makes the one optional. So now we only collect 10 digits and append the on in the bridge line. /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: > I'm sorry, but do not understand what it is that I should try. Are you > saying to change the data attribute in the action command of the > dialplan? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/1f9f0771/attachment.html From larclap at yahoo.com Mon May 11 16:51:44 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 11 May 2009 16:51:44 -0700 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> Message-ID: <017601c9d293$709358e0$51ba0aa0$@com> Thanks for the clarification. It made sense, but the results remain the same. The log still says 'Invalid Gateway' and 'sofia status' at the console does not show flowroute. Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Regex (PASS) [Long Distance - flowroute] destination_number(3235551212) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action set(effective_caller_id_number=13235555555) Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action bridge(sofia/gateway/flowroute/flowrouteAccount#13235551212) 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1000 at 192.168.10.29:5090) State Change CS_ROUTING -> CS_EXECUTE 2009-05-11 16:43:47 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at 192.168.10.29:5090 [BREAK] 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) State ROUTING going to sleep 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) Running State Change CS_EXECUTE 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) State EXECUTE 2009-05-11 16:43:47 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1000 at 192.168.10.29:5090 SOFIA EXECUTE 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1000 at 192.168.10.29:5090 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(open=true) 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(use_profile=nat) 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [use_profile]=[nat] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set_user(default at 192.168.10.29) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/192.168.10.29-spymap/1000/924426fa-3e85-11de-9b6b-7ba002f89c82) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/192.168.10.29-last_dial/1000/3235551212) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/192.168.10.29-last_dial/global/924426fa-3e85-11de-9b6b-7ba002f89 c82) EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(effective_caller_id_number=13235555555) 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [effective_caller_id_number]=[13235555555] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(effective_caller_id_name=FreeSWITCH) 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29:5090 SET [effective_caller_id_name]=[FreeSWITCH] EXECUTE sofia/internal/1000 at 192.168.10.29:5090 bridge(sofia/gateway/flowroute/flowrouteAccount#13235551212) 2009-05-11 16:43:47 [ERR] mod_sofia.c:2531 sofia_outgoing_channel() Invalid Gateway 2009-05-11 16:43:47 [NOTICE] mod_sofia.c:2744 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (N/A) State DESTROY 2009-05-11 16:43:47 [DEBUG] mod_sofia.c:240 sofia_on_destroy() N/A SOFIA DESTROY 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (N/A) State DESTROY going to sleep 2009-05-11 16:43:47 [ERR] switch_ivr_originate.c:1494 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, May 11, 2009 3:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't configure outbound call You want to collect 11 or 10 digits and send it out flowroute which requires 1+ number. The 1? makes the one optional. So now we only collect 10 digits and append the on in the bridge line. /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: I'm sorry, but do not understand what it is that I should try. Are you saying to change the data attribute in the action command of the dialplan? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/0eb0d7bc/attachment-0001.html From dftoro at yahoo.com Mon May 11 16:56:25 2009 From: dftoro at yahoo.com (Diego Toro) Date: Mon, 11 May 2009 16:56:25 -0700 (PDT) Subject: [Freeswitch-users] get call durantion Message-ID: <211281.48070.qm@web33508.mail.mud.yahoo.com> Hi, ? I need get call duration after bridge application using mod_managed, my code: ?Session.Execute("bridge", sbNewOutBoundNum); ??? ?? Debug("billsec :" + _Session.GetVariable("billsec")); ?? Debug("duration :" + _Session.GetVariable("duration")); ? The bridge is ok, but?the variable value duration?and?billsec is zero (0). ? Diego --- On Sun, 5/10/09, Michael S Collins wrote: Do you mean from the CDR? I recommend XML CDRs because they give tons of information. If you are talking about gathering this stuff midcall then you'll need to supply more information about your setup.? -MC Sent from my iPhone On May 10, 2009, at 6:17 PM, Diego Toro wrote: Hi, How can I get call durantion after bridge application ? ?I tried with billsec and duration but I don't get any value. ? Thank you _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/409878a6/attachment.html From brian at freeswitch.org Mon May 11 16:57:53 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 18:57:53 -0500 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <017601c9d293$709358e0$51ba0aa0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> <017601c9d293$709358e0$51ba0aa0$@com> Message-ID: <303E743A-BE63-46FD-9D46-15BB0F227E4D@freeswitch.org> Please contact your provider for assistance. I can't guess what they might want! /b On May 11, 2009, at 6:51 PM, Lars Zeb wrote: > Thanks for the clarification. It made sense, but the results remain > the same. The log still says ?Invalid Gateway? and ?sofia status? at > the console does not show flowroute. > > Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Regex (PASS) [Long > Distance - flowroute] destination_number(3235551212) =~ /^1?(\d{10}) > $/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action > set(effective_caller_id_number=13235555555) > Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action > set(effective_caller_id_name=${outbound_caller_id_name}) > Dialplan: sofia/internal/1000 at 192.168.10.29:5090 Action bridge(sofia/ > gateway/flowroute/flowrouteAccount#13235551212) > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:114 > switch_core_standard_on_routing() (sofia/internal/ > 1000 at 192.168.10.29:5090) State Change CS_ROUTING -> CS_EXECUTE > 2009-05-11 16:43:47 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal sofia/internal/1000 at 192.168.10.29 > :5090 [BREAK] > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) > State ROUTING going to sleep > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) > Running State Change CS_EXECUTE > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/internal/1000 at 192.168.10.29:5090) > State EXECUTE > 2009-05-11 16:43:47 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1000 at 192.168.10.29 > :5090 SOFIA EXECUTE > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:151 > switch_core_standard_on_execute() sofia/internal/ > 1000 at 192.168.10.29:5090 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(open=true) > 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29 > :5090 SET [open]=[true] > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set(use_profile=nat) > 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29 > :5090 SET [use_profile]=[nat] > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 set_user(default at 192.168.10.29 > ) > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/ > 192.168.10.29-spymap/1000/924426fa-3e85-11de-9b6b-7ba002f89c82) > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/ > 192.168.10.29-last_dial/1000/3235551212) > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 hash(insert/ > 192.168.10.29-last_dial/global/924426fa-3e85-11de-9b6b-7ba002f89c82) > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 > set(effective_caller_id_number=13235555555) > 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29 > :5090 SET [effective_caller_id_number]=[13235555555] > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 > set(effective_caller_id_name=FreeSWITCH) > 2009-05-11 16:43:47 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1000 at 192.168.10.29 > :5090 SET [effective_caller_id_name]=[FreeSWITCH] > EXECUTE sofia/internal/1000 at 192.168.10.29:5090 bridge(sofia/gateway/ > flowroute/flowrouteAccount#13235551212) > 2009-05-11 16:43:47 [ERR] mod_sofia.c:2531 sofia_outgoing_channel() > Invalid Gateway > 2009-05-11 16:43:47 [NOTICE] mod_sofia.c:2744 > sofia_outgoing_channel() Close Channel N/A [CS_NEW] > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (N/A) State DESTROY > 2009-05-11 16:43:47 [DEBUG] mod_sofia.c:240 sofia_on_destroy() N/A > SOFIA DESTROY > 2009-05-11 16:43:47 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (N/A) State DESTROY going to sleep > 2009-05-11 16:43:47 [ERR] switch_ivr_originate.c:1494 > switch_ivr_originate() Cannot create outgoing channel of type > [sofia] cause: [INVALID_NUMBER_FORMAT] > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/b157406d/attachment-0001.html From fdhege at gmail.com Mon May 11 17:08:01 2009 From: fdhege at gmail.com (Dale) Date: Mon, 11 May 2009 20:08:01 -0400 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <017601c9d293$709358e0$51ba0aa0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> <017601c9d293$709358e0$51ba0aa0$@com> Message-ID: <69063608-12F5-4724-BBFF-EA1060662373@gmail.com> On May 11, 2009, at 7:51 PM, Lars Zeb wrote: > Thanks for the clarification. It made sense, but the results remain > the same. The log still says ?Invalid Gateway? and ?sofia status? at > the console does not show flowroute. It sounds like sofia hasn't picked up your new gateway yet. Have you tried something like the below yet? sofia profile external rescan reloadxml -Dale -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/9fea6baa/attachment.html From cliff at develix.com Mon May 11 17:22:41 2009 From: cliff at develix.com (Cliff Wells) Date: Mon, 11 May 2009 17:22:41 -0700 Subject: [Freeswitch-users] rdnis variable from Lua Message-ID: <1242087761.6170.18.camel@portable-evil> Hi, I can see the RDN in the log file, but don't know how to retrieve it from a Lua script. Regards, Cliff From cliff at develix.com Mon May 11 18:20:54 2009 From: cliff at develix.com (Cliff Wells) Date: Mon, 11 May 2009 18:20:54 -0700 Subject: [Freeswitch-users] rdnis variable from Lua In-Reply-To: <1242087761.6170.18.camel@portable-evil> References: <1242087761.6170.18.camel@portable-evil> Message-ID: <1242091254.6170.21.camel@portable-evil> I found a workaround, but it'd be nice to actually have the RDN easily accessible from Lua: calling_number = session:getVariable ( "sip_h_Diversion" ) _, _, calling_number = string.find ( calling_number, "sip:(%d+)@" ) Cliff On Mon, 2009-05-11 at 17:22 -0700, Cliff Wells wrote: > Hi, > > I can see the RDN in the log file, but don't know how to retrieve it > from a Lua script. > > Regards, > Cliff > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon May 11 19:59:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 21:59:16 -0500 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <014e01c9d287$27d5a2e0$7780e8a0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> Message-ID: The one you emailed anthony about the invite... 487 repeat stuff.. can you give me more details on what might be going on? /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: > I'm sorry, but do not understand what it is that I should try. Are you > saying to change the data attribute in the action command of the > dialplan? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/71f54da0/attachment.html From brian at freeswitch.org Mon May 11 20:00:50 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 22:00:50 -0500 Subject: [Freeswitch-users] rdnis variable from Lua In-Reply-To: <1242091254.6170.21.camel@portable-evil> References: <1242087761.6170.18.camel@portable-evil> <1242091254.6170.21.camel@portable-evil> Message-ID: <24F9155C-4467-497D-9B81-6960E1669016@freeswitch.org> Hehe its not really a work around... Its how you do it either way... but I did add the patch from http://jira.freeswitch.org/browse/MODSOFIA-7 which would require you to do similar. /b On May 11, 2009, at 8:20 PM, Cliff Wells wrote: > I found a workaround, but it'd be nice to actually have the RDN easily > accessible from Lua: > > calling_number = session:getVariable ( "sip_h_Diversion" ) > _, _, calling_number = string.find ( calling_number, "sip:(%d+)@" ) > > > Cliff Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/c7553bae/attachment.html From juanma.v82 at gmail.com Mon May 11 20:03:44 2009 From: juanma.v82 at gmail.com (Juan Manuel Vicente) Date: Tue, 12 May 2009 00:03:44 -0300 Subject: [Freeswitch-users] SDP Passthrough, INVITE messages. Message-ID: <26613bbf0905112003m2233b98ax26c8c4f6f6091b22@mail.gmail.com> Hi, I'm trying to use the Freeswitch as a proxy (I know that is not designed for that, but I really need to do it in this way), here is my config: Endpoint 1<-> FS A-<->FS B<->FS A<->Endpoint 2 * Both Endpoints are registered in FS A how is acting as a proxy and registrar. * FS B only sends back the Invite to FS A in order to reach Endpoint 2. * Both FS have a public IP * FS A Only handles SIP messages * FS B Handles RTP (Also SIP) My objetive is to keep the signaling in FS A and the RTP in FS B so basically FS A will work as a registrar. So far I've been able to succesfully do it if both endpoint are not nated, how ever I do need to do it in a Natted sceneario too, for what I have been sniffing the problem is that in the INVITE, the SDP is sending the internal IP instead of the external. I've tried to change the switch_r_sdp and switch_l_sdp but I'm not quite sure if I'm doing the correct config of the switch (late_codec_negotiation) If anyone could give a tip or a sample of how can I change the INVITE messages I will appreciate. Thanks in advance Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/1b0f5aad/attachment.html From brian at freeswitch.org Mon May 11 20:11:31 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 22:11:31 -0500 Subject: [Freeswitch-users] SDP Passthrough, INVITE messages. In-Reply-To: <26613bbf0905112003m2233b98ax26c8c4f6f6091b22@mail.gmail.com> References: <26613bbf0905112003m2233b98ax26c8c4f6f6091b22@mail.gmail.com> Message-ID: <5FC818CE-80B1-4211-A4F6-DAFE110FF575@freeswitch.org> Juan, Can you explain your situation a little better you seem to have breezed over the critical details. Also you should enable STUN on your endpoints and not depend on your Registrar to overcome nat issues since its not its job. /b On May 11, 2009, at 10:03 PM, Juan Manuel Vicente wrote: > So far I've been able to succesfully do it if both endpoint are not > nated, how ever I do need to do it in a Natted sceneario too, for > what I have been sniffing the problem is that in the INVITE, the SDP > is sending the internal IP instead of the external. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/8e66337a/attachment-0001.html From larclap at yahoo.com Mon May 11 21:29:36 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 11 May 2009 21:29:36 -0700 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> Message-ID: <01b401c9d2ba$41d09af0$c571d0d0$@com> I believe you're talking about the FS sending out an ICMP in response to the client's invite, which resulted in 'ICMP Destination unreachable'. He told me to turn iptables off, which I did. Then the Eyebeam registered successfully. I don't know what the '487 repeat stuff' was. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, May 11, 2009 7:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Can't configure outbound call The one you emailed anthony about the invite... 487 repeat stuff.. can you give me more details on what might be going on? /b On May 11, 2009, at 5:23 PM, Lars Zeb wrote: I'm sorry, but do not understand what it is that I should try. Are you saying to change the data attribute in the action command of the dialplan? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/011c1b57/attachment.html From brian at freeswitch.org Mon May 11 21:58:38 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 11 May 2009 23:58:38 -0500 Subject: [Freeswitch-users] Can't configure outbound call In-Reply-To: <01b401c9d2ba$41d09af0$c571d0d0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <01b401c9d2ba$41d09af0$c571d0d0$@com> Message-ID: <25F42139-B287-42D6-9907-F9CFA81014E5@freeswitch.org> Haha that's ok I sent that email to you by mistake .... I wondered where that email went! /b Sent from my iPhone On May 11, 2009, at 11:29 PM, "Lars Zeb" wrote: > I believe you?re talking about the FS sending out an ICMP in respons > e to the client?s invite, which resulted in ?ICMP Destination > unreachable?. He told me to turn iptables off, which I did. Then th > e Eyebeam registered successfully. I don?t know what the ?487 > repeat stuff? was. > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: Monday, May 11, 2009 7:59 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Can't configure outbound call > > > > The one you emailed anthony about the invite... 487 repeat stuff.. > can you give me more details on what might be going on? > > > > /b > > > > On May 11, 2009, at 5:23 PM, Lars Zeb wrote: > > > > > I'm sorry, but do not understand what it is that I should try. Are you > saying to change the data attribute in the action command of the > dialplan? > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090511/a12cc8a7/attachment.html From mrene_lists at avgs.ca Mon May 11 22:35:26 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 12 May 2009 01:35:26 -0400 Subject: [Freeswitch-users] get call durantion In-Reply-To: <211281.48070.qm@web33508.mail.mud.yahoo.com> References: <211281.48070.qm@web33508.mail.mud.yahoo.com> Message-ID: Those variables are only set on hangup. You can have a look at the caller profile to get the answered time if you want to process that before hangup. Math On 11-May-09, at 7:56 PM, Diego Toro wrote: > Hi, > > I need get call duration after bridge application using mod_managed, > my code: > > Session.Execute("bridge", sbNewOutBoundNum); > > > Debug("billsec :" + _Session.GetVariable("billsec")); > Debug("duration :" + _Session.GetVariable("duration")); > > The bridge is ok, but the variable value duration and billsec is > zero (0). > > Diego > > --- On Sun, 5/10/09, Michael S Collins wrote: > > > > Do you mean from the CDR? I recommend XML CDRs because they give > tons of information. If you are talking about gathering this stuff > midcall then you'll need to supply more information about your setup. > > -MC > > Sent from my iPhone > > On May 10, 2009, at 6:17 PM, Diego Toro wrote: > >> Hi, >> How can I get call durantion after bridge application ? I tried >> with billsec and duration but I don't get any value. >> >> Thank you >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/ec0c9e64/attachment-0001.html From diego.viola at gmail.com Mon May 11 22:45:22 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 12 May 2009 01:45:22 -0400 Subject: [Freeswitch-users] get call durantion In-Reply-To: References: <211281.48070.qm@web33508.mail.mud.yahoo.com> Message-ID: <86a32abc0905112245p4b400533xe1c881458196e7ed@mail.gmail.com> You can do /event list all on the CLI, and on hangup you will see that the billsec, duration and other variables are set. Regards, Diego On Tue, May 12, 2009 at 1:35 AM, Mathieu Rene wrote: > Those variables are only set on hangup. You can have a look at the caller > profile to get the answered time if you want to process that before hangup. > Math > On 11-May-09, at 7:56 PM, Diego Toro wrote: > > Hi, > > I need get call duration after bridge application using mod_managed, my > code: > ?Session.Execute("bridge", sbNewOutBoundNum); > > > ?? Debug("billsec :" + _Session.GetVariable("billsec")); > ?? Debug("duration :" + _Session.GetVariable("duration")); > > The bridge is ok, but?the variable value duration?and?billsec is zero (0). > > Diego > > --- On Sun, 5/10/09, Michael S Collins wrote: > > > > Do you mean from the CDR? I recommend XML CDRs because they give tons of > information. If you are talking about gathering this stuff midcall then > you'll need to supply more information about your setup. > -MC > > Sent from my iPhone > On May 10, 2009, at 6:17 PM, Diego Toro wrote: > > Hi, > How can I get call durantion after bridge application ? ?I tried with > billsec and duration but I don't get any value. > > Thank you > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From peter.olsson at visionutveckling.se Tue May 12 02:36:02 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 12 May 2009 11:36:02 +0200 Subject: [Freeswitch-users] Audio "clicks" between playback of audio files In-Reply-To: <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363906@cooper> Brian, After converting all the files to .wav it works great. And it also solved my issue for recording messages. This is good enough for me - even though I guess there might be some issues with native files i FreeSWITCH. Thanks for all your help! /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 11 maj 2009 16:24 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Audio "clicks" between playback of audio files I'm not sure. Can you provide me a test file and a known case that you can produce this issue with? /b On May 11, 2009, at 9:20 AM, Peter Olsson wrote: Brian, Thanks for the response. No, I didn't try wav files - and I'd prefer to keep the current codec if that's possible. But I could give it a try and see what happens. Do you think it might only be related to the native files in FS? //Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a08362532934700715407! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/fb73ac43/attachment.html From ibrahim.tunali at comturk.com Tue May 12 00:11:31 2009 From: ibrahim.tunali at comturk.com (=?ISO-8859-9?Q?=DDbrahim_TUNALI?=) Date: Tue, 12 May 2009 10:11:31 +0300 Subject: [Freeswitch-users] mod_event_socket 183 early media Message-ID: Hi, I am trying to build a SBC but a kind of different described on wiki. I immediately send incoming leg A to a socket app and connect bridge with my socket python application. I will handle all business logic on python side. My dialplan is so simple; Freeswitch send calls to parking and 183 early media through mod_sofia. This is unacceptable on my SBC scenario. Because I not receive 183 or 180 from leg B yet. I want to relay early media or ringing when I received from B leg I realize this is not a bug. It is a feature but i need to know is this possible. I would appreciate all suggestions and clues. Regards, ibrahim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/f05d5447/attachment.html From ibrahim.tunali at comturk.com Tue May 12 04:33:49 2009 From: ibrahim.tunali at comturk.com (=?ISO-8859-9?Q?=DDbrahim_TUNALI?=) Date: Tue, 12 May 2009 14:33:49 +0300 Subject: [Freeswitch-users] mod_event_socket 183 early media Message-ID: Hi, I am trying to build a SBC but a kind of different described on wiki. I immediately send incoming leg A to a socket app and connect bridge with my socket python application. I will handle all business logic on python side. My dialplan is so simple; Freeswitch send calls to parking and 183 early media through mod_sofia. This is unacceptable on my SBC scenario. Because I not receive 183 or 180 from leg B yet. I want to relay early media or ringing when I received from B leg I realize this is not a bug. It is a feature but i need to know is this possible. I would appreciate all suggestions and clues. Regards, ibrahim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/390196b1/attachment.html From niall.crosby at gmail.com Tue May 12 01:10:01 2009 From: niall.crosby at gmail.com (Ceolter) Date: Tue, 12 May 2009 01:10:01 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Java Socket Interface API Message-ID: <23462633.post@talk.nabble.com> Hi Klaus, I was wondering if you are still working on the Java Socket Interface for Freeswitch. I'd be interested in using it if it exists. Thanks, Niall. Klaus L'Imbecile wrote: > > Hi Folks, > > Just to let you know that we are working on a library for connecting to > the Freeswitch via the socket interface. We plan to release it under LGPL > as soon as it's somewhat robust. > > > And for those of you who are in the USA, please don't forget to go vote. > > Thanks, > Klaus. > -- > "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ... > Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-Java-Socket-Interface-API-tp20303263p23462633.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Tue May 12 08:00:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 12 May 2009 10:00:12 -0500 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: References: Message-ID: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> the socket app will always open early media, it's not currently possible to park a channel that does not have media, you could try to post a bounty for such a feature but it would have to be an elegant solution 2009/5/12 ?brahim TUNALI > Hi, > I am trying to build a SBC but a kind of different described on wiki. > > I immediately send incoming leg A to a socket app and connect bridge with > my socket python application. I will handle all business logic on python > side. > > My dialplan is so simple; > > > > > > > > Freeswitch send calls to parking and 183 early media through mod_sofia. > This is unacceptable on my SBC scenario. Because I not receive 183 or 180 > from leg B yet. I want to relay early media or ringing when I received from > B leg > > I realize this is not a bug. It is a feature but i need to know is this > possible. > > I would appreciate all suggestions and clues. > > Regards, > ibrahim > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/aefdbce8/attachment-0001.html From dave at 3c.co.uk Tue May 12 10:49:24 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 12 May 2009 18:49:24 +0100 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> Message-ID: <1242150564.4019.6.camel@dk-d820> This is something I've been wondering about as well. What's the reason for the channel being parked? Cheers -- Dave > the socket app will always open early media, it's not currently > possible to park a channel that does not have media, you could try to > post a bounty for such a feature but it would have to be an elegant > solution > > > 2009/5/12 ?brahim TUNALI > Hi, > I am trying to build a SBC but a kind of different described > on wiki. > > I immediately send incoming leg A to a socket app and connect > bridge with my socket python application. I will handle all > business logic on python side. > > My dialplan is so simple; > > > expression="^1000$"> > > > > > Freeswitch send calls to parking and 183 early media through > mod_sofia. This is unacceptable on my SBC scenario. Because I > not receive 183 or 180 from leg B yet. I want to relay early > media or ringing when I received from B leg > > I realize this is not a bug. It is a feature but i need to > know is this possible. > > I would appreciate all suggestions and clues. > > Regards, > ibrahim > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue May 12 11:00:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 May 2009 11:00:37 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: www.freeswitch.org site updated Message-ID: <87f2f3b90905121100m4346bb76q7e12f2e415cf0908@mail.gmail.com> FYI, We'd like to let the community know that we've moved the main site, www.freeswitch.org, to a new server and have updated it to the latest Drupal version. We are still calibrating the server so if the site seems slower than usual please let me know off list and we'll take a look. Also, because this upgrade was so major we were forced to wipe out all the users from the previous Drupal install. Please feel free to sign up again if you so desire. You only need to sign up if you wish to post comments. One other new feature is the "Share This" button. (You need JavaScript enabled to see this.) It makes it as easy as possible to share stories with Digg, SU, Deli.ico.us, etc. If there are any stories you've not already dugg then please try out the Share This button to see how easy it is to digg stuff now. As always, if you have any questions or comments then please email me off list and I'll address them quickly. -Michael http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/9cff9db3/attachment.html From anthony.minessale at gmail.com Tue May 12 11:13:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 12 May 2009 13:13:42 -0500 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <1242150564.4019.6.camel@dk-d820> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> Message-ID: <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> The park state is where the channel goes to be kept alive while you hold the socket. The channel has to be in constant motion reading and writing audio or the audio would build up, akin to a gui app blocking and you can't press any of the buttons. it may be possible for the park loop to tighten up when the call is not answered or in early media in advance but it would not be able to do anything besides a few things like get and set variables until it was meda-enabled On Tue, May 12, 2009 at 12:49 PM, David Knell wrote: > This is something I've been wondering about as well. What's the reason > for the channel being parked? > > Cheers -- > > Dave > > > the socket app will always open early media, it's not currently > > possible to park a channel that does not have media, you could try to > > post a bounty for such a feature but it would have to be an elegant > > solution > > > > > > 2009/5/12 ?brahim TUNALI > > Hi, > > I am trying to build a SBC but a kind of different described > > on wiki. > > > > I immediately send incoming leg A to a socket app and connect > > bridge with my socket python application. I will handle all > > business logic on python side. > > > > My dialplan is so simple; > > > > > > > expression="^1000$"> > > > > > > > > > > Freeswitch send calls to parking and 183 early media through > > mod_sofia. This is unacceptable on my SBC scenario. Because I > > not receive 183 or 180 from leg B yet. I want to relay early > > media or ringing when I received from B leg > > > > I realize this is not a bug. It is a feature but i need to > > know is this possible. > > > > I would appreciate all suggestions and clues. > > > > Regards, > > ibrahim > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/79e8636d/attachment.html From msc at freeswitch.org Tue May 12 11:24:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 May 2009 11:24:37 -0700 Subject: [Freeswitch-users] Cluecon 2009 - Hotel Special! Message-ID: <87f2f3b90905121124s7e4c8d22ye3c6d287834acfd4@mail.gmail.com> Good news! For those of you who like to get a good deal on a hotel room we've got the inside scoop on a special for the Wyndham Chicago. I just did a search on expedia.com for a hotel with these specs: Hotel only Check-in: 8/3/2009, Check-Out: 8/6/2009 1 room, 2 adults City: Chicago Hotel Name: "Wyndham Chicago" The results were very nice: the non-refundable rate per night is only $140!!! Get to expedia.com NOW and get signed up! We look forward to seeing you in Chicago this August. -Michael http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/b3c3ca4b/attachment.html From juanma.v82 at gmail.com Tue May 12 11:59:57 2009 From: juanma.v82 at gmail.com (Juan Manuel Vicente) Date: Tue, 12 May 2009 15:59:57 -0300 Subject: [Freeswitch-users] SDP Passthrough, INVITE messages. In-Reply-To: <5FC818CE-80B1-4211-A4F6-DAFE110FF575@freeswitch.org> References: <26613bbf0905112003m2233b98ax26c8c4f6f6091b22@mail.gmail.com> <5FC818CE-80B1-4211-A4F6-DAFE110FF575@freeswitch.org> Message-ID: <26613bbf0905121159n595a0ca5oc03f38ad9388f59a@mail.gmail.com> Brian, thanks for your reply I going to try to explain better my problem. I will use examples to do this task. Scenario A: Endpoint 1 <--(sip,rtp)--> FS <--(sip,rtp)--> Endpoint 2 * Endpoint 1 is nated. * Endpoint 2 is not nated * Endpoint 1 is sending wrong SDP info, sending local IP instead of public IP (switch_r_sdp). but FS can fix this problem with the switch_r_sdp in this scenario is: "[v=0 o=- 3 2 IN IP4 10.0.100.21 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.100.21 t=0 0 m=audio 7208 RTP/AVP 100 106 0 105 98 8 3 101 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : rnYgTtVR uOhvAtG7 10.0.100.21 7208 ]" Scenario B: SIP: Endpoint 1 <--(sip)--> FS A <--(sip)--> Endpoint 2 RTP Endpoint 1 <--(rtp)--> FS A <--(rtp)--> Endpoint 2 * Endpoint 1 is nated. * Endpoint 2 is not nated * Fs is configured with the switch_r_sdp in this scenario is: "[v=0 o=- 3 2 IN IP4 10.0.100.21 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.100.21 t=0 0 m=audio 7208 RTP/AVP 100 106 0 105 98 8 3 101 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : rnYgTtVR uOhvAtG7 10.0.100.21 7208 ]" But it is wrong, the correct one must be: [v=0 o=- 3 2 IN IP4 200.49.24.35 s=CounterPath X-Lite 3.0 c=IN IP4 200.49.24.35 t=0 0 m=audio 63936 RTP/AVP 100 106 0 105 98 8 3 101 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : CtEyvKrr I3ZLmyBD 10.0.100.21 63936 a=alt:2 1 : 6JO5oo5+ hWtLJ+lx 200.49.24.35 63936 ] To have a correct one I had to configure stun in the endpoint, but the problem is my infrastructure is close to 1000 Endpoints and in some cases I cant reach then to configure properly. There any way to fix switch_r_sdp? I can use Opensips or similar SIP proxy but this can add another possible point to failure, and the issue is quite simple. Thanks in Advance. On Tue, May 12, 2009 at 12:11 AM, Brian West wrote: > Juan, Can you explain your situation a little better you seem to have > breezed over the critical details. Also you should enable STUN on your > endpoints and not depend on your Registrar to overcome nat issues since its > not its job. > > /b > > On May 11, 2009, at 10:03 PM, Juan Manuel Vicente wrote: > > So far I've been able to succesfully do it if both endpoint are not nated, > how ever I do need to do it in a Natted sceneario too, for what I have been > sniffing the problem is that in the INVITE, the SDP is sending the internal > IP instead of the external. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090512/e4721ae5/attachment-0001.html From gmaruzz at celliax.org Tue May 12 12:24:17 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 12 May 2009 21:24:17 +0200 Subject: [Freeswitch-users] Cluecon 2009 - Hotel Special! In-Reply-To: <87f2f3b90905121124s7e4c8d22ye3c6d287834acfd4@mail.gmail.com> References: <87f2f3b90905121124s7e4c8d22ye3c6d287834acfd4@mail.gmail.com> Message-ID: <7b197bef0905121224u126caa5lb88ea5a56eef45f6@mail.gmail.com> yay Michael! Got it! I owe you one!!!!! On Tue, May 12, 2009 at 8:24 PM, Michael Collins wrote: > Good news! For those of you who like to get a good deal on a hotel room > we've got the inside scoop on a special for the Wyndham Chicago. I just did > a search on expedia.com for a hotel with these specs: > Hotel only > Check-in: 8/3/2009, Check-Out: 8/6/2009 > 1 room, 2 adults > City: Chicago > Hotel Name: "Wyndham Chicago" > > The results were very nice: the non-refundable rate per night is only > $140!!! Get to expedia.com NOW and get signed up! > > We look forward to seeing you in Chicago this August. > -Michael > http://www.cluecon.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dave at 3c.co.uk Tue May 12 16:23:12 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 13 May 2009 00:23:12 +0100 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> Message-ID: <1242170592.4019.13.camel@dk-d820> Gotcha - but in the case where the call hasn't yet got to a point where there's a 183 been sent then I guess this wouldn't apply - there shouldn't be any audio from the far end at this point, nor would the far end be expecting any. I'd suggest (and would volunteer to knock together a patch) adding a 'nopark' option to the socket command, which doesn't park the call - nor would it change existing behaviour. Obviously, in a situation like that outlined by Ibrahim where the socket app handles all aspects of the call, then it'll need to make sure that it signals ringing, answer or whatever to make the call state flow work. --Dave > The park state is where the channel goes to be kept alive while you > hold the socket. > > The channel has to be in constant motion reading and writing audio or > the audio would build up, akin to a gui app blocking and you can't > press any of the buttons. > > it may be possible for the park loop to tighten up when the call is > not answered or in early media in advance > but it would not be able to do anything besides a few things like get > and set variables until it was meda-enabled > > > On Tue, May 12, 2009 at 12:49 PM, David Knell wrote: > This is something I've been wondering about as well. What's > the reason > for the channel being parked? > > Cheers -- > > Dave > > > > the socket app will always open early media, it's not > currently > > possible to park a channel that does not have media, you > could try to > > post a bounty for such a feature but it would have to be an > elegant > > solution > > > > > > 2009/5/12 ?brahim TUNALI > > Hi, > > I am trying to build a SBC but a kind of different > described > > on wiki. > > > > I immediately send incoming leg A to a socket app > and connect > > bridge with my socket python application. I will > handle all > > business logic on python side. > > > > My dialplan is so simple; > > > > > > > expression="^1000$"> > > data="127.0.0.1:1905 > > async full"/> > > > > > > > > Freeswitch send calls to parking and 183 early media > through > > mod_sofia. This is unacceptable on my SBC scenario. > Because I > > not receive 183 or 180 from leg B yet. I want to > relay early > > media or ringing when I received from B leg > > > > I realize this is not a bug. It is a feature but i > need to > > know is this possible. > > > > I would appreciate all suggestions and clues. > > > > Regards, > > ibrahim > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ibrahim.tunali at comturk.com Wed May 13 00:12:00 2009 From: ibrahim.tunali at comturk.com (ibrahim tunali) Date: Wed, 13 May 2009 00:12:00 -0700 (PDT) Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <1242170592.4019.13.camel@dk-d820> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> Message-ID: <23516585.post@talk.nabble.com> "nopark" option would be great. FS sends "100 trying" while opening socket and leg A waits other responses from socket. If I send pre-answer command it reply via 183 early media and activate RTP path. Regards, ibrahim David Knell wrote: > > Gotcha - but in the case where the call hasn't yet got to a point where > there's a 183 been sent then I guess this wouldn't apply - there > shouldn't be any audio from the far end at this point, nor would the far > end be expecting any. > > I'd suggest (and would volunteer to knock together a patch) adding a > 'nopark' option to the socket command, which doesn't park the call - nor > would it change existing behaviour. Obviously, in a situation like that > outlined by Ibrahim where the socket app handles all aspects of the > call, then it'll need to make sure that it signals ringing, answer or > whatever to make the call state flow work. > > --Dave > > -- View this message in context: http://www.nabble.com/mod_event_socket-183-early-media-tp23503263p23516585.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mikael at bjerkeland.com Wed May 13 00:30:27 2009 From: mikael at bjerkeland.com (Mikael Aleksander Bjerkeland) Date: Wed, 13 May 2009 09:30:27 +0200 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <23516585.post@talk.nabble.com> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> <23516585.post@talk.nabble.com> Message-ID: <1242199827.10932.1.camel@mikael-xpsm1530> I've been looking for something like this as well, to make the outbound event socket behave more like FastAGI and handle the logic. El mi?, 13-05-2009 a las 00:12 -0700, ibrahim tunali escribi?: > "nopark" option would be great. FS sends "100 trying" while opening socket > and leg A waits other responses from socket. If I send pre-answer command it > reply via 183 early media and activate RTP path. > > Regards, > ibrahim > > > David Knell wrote: > > > > Gotcha - but in the case where the call hasn't yet got to a point where > > there's a 183 been sent then I guess this wouldn't apply - there > > shouldn't be any audio from the far end at this point, nor would the far > > end be expecting any. > > > > I'd suggest (and would volunteer to knock together a patch) adding a > > 'nopark' option to the socket command, which doesn't park the call - nor > > would it change existing behaviour. Obviously, in a situation like that > > outlined by Ibrahim where the socket app handles all aspects of the > > call, then it'll need to make sure that it signals ringing, answer or > > whatever to make the call state flow work. > > > > --Dave > > > > > From helmut.kuper at ewetel.de Wed May 13 00:55:36 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 13 May 2009 09:55:36 +0200 Subject: [Freeswitch-users] RFC: pickup using a two nodes FS PBX Message-ID: <4A0A7CF8.7040809@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I want to allow pickup in a two nodes/servers FS PBX sharing the same database. So phone A on node 1 rings phone B on node 1. phone C on node 2 tries to pickup phone A. mod_limit is working on both nodes sharing the same database. So pickup function in node 2 sees the incomming call from phone A and knows the corresponding uuid. Can I use such functions like intercept in this multi server context? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCnz44tZeNddg3dwRAhCOAKCq7H7OyXvIRivnWjKYw6IaKzQCCgCfQQQ0 rWLR9w1lJ4CrPj+nu0cV6EE= =jQwK -----END PGP SIGNATURE----- From yivzhenko at mksat.net Wed May 13 03:32:35 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Wed, 13 May 2009 13:32:35 +0300 Subject: [Freeswitch-users] Set nibblebill variables on B-leg Message-ID: <200905131332.35250.yivzhenko@mksat.net> Hi All, I want to use mod_nibblebill for billing outgoing calls. My problem is billing begins before other paty answers the call. The WiKi tells: "You need to set the billing variables on your outbound calling leg and NOT on your A-Leg." How i can set nibblebill variables when B-leg connects? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/6730b3e6/attachment.html From saeedahmad1981 at gmail.com Wed May 13 04:40:23 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 13 May 2009 13:40:23 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 Message-ID: <2059908291DA4436B2009BBB52584C46@saeedlaptop> Hi, I just did 'make current' and FS failed to load xml_curl, I looked into mod/ dir but mod_xml_curl.so wasn't there. I even tried to copy it from src directory but somehow FS doesn't load it. What could be the problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/73b857e1/attachment.html From tristan at telemaque.fr Wed May 13 05:15:11 2009 From: tristan at telemaque.fr (Tristan) Date: Wed, 13 May 2009 14:15:11 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <2059908291DA4436B2009BBB52584C46@saeedlaptop> References: <2059908291DA4436B2009BBB52584C46@saeedlaptop> Message-ID: <4A0AB9CF.4030307@telemaque.fr> Hi Saeed, Maybe you did not built it ? ( see in modules.conf if it's not commented out with a # ) Saeed Ahmed a ?crit : > > Hi, > > I just did 'make current' and FS failed to load xml_curl, I looked > into mod/ dir but mod_xml_curl.so wasn't there. I even tried to copy > it from src directory but somehow FS doesn't load it. What could be > the problem? > > Thanks > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/bc9c3ec2/attachment-0001.html From odermann at googlemail.com Wed May 13 05:22:38 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 13 May 2009 14:22:38 +0200 Subject: [Freeswitch-users] Fax through FS to Callweaver. How? Message-ID: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> hi, we are doing a lot with freeswitch and everything works absolutely perfect. the only thing we are missing, is a good support for fax over ip - may it be for offering a fax-2-mail service or just passing incoming fax-calls from our fs-servers to a "real" fax-machine. now we have another server only for callweaver, which has a great fax-support. if we send faxes from our fax-machine over our sip-provider (who supports t.38) directly to the cw-server, all faxes are beeing received and saved by cw. but if we send faxes over the same sip-provider first to fs and then do a bridge to the cw-server, all faxes fail. we set in internal.xml and play with bypass_media=true and/or proxy_media=true in the dialplan. it simply does not work. i think and hope that there are some other settings i can change, so that fs is nothing more than just a "tunnel" and let both sides communicate with eachother? does someone know callweaver and can tell me, if there are some important settings to be set for making it work with fs in the middle? thanks & kind regards dennis From saeedahmad1981 at gmail.com Wed May 13 05:56:46 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 13 May 2009 14:56:46 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <4A0AB9CF.4030307@telemaque.fr> References: <2059908291DA4436B2009BBB52584C46@saeedlaptop> <4A0AB9CF.4030307@telemaque.fr> Message-ID: <2F7A3C109BB644DAAD92816534F1948F@saeedlaptop> Hi Tristan, No its not commented out. And strange thing is that it was working until I did ?make current? today. Thanks for your response. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tristan Sent: Wednesday, May 13, 2009 2:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 Hi Saeed, Maybe you did not built it ? ( see in modules.conf if it's not commented out with a # ) Saeed Ahmed a ?crit : Hi, I just did ?make current? and FS failed to load xml_curl, I looked into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy it from src directory but somehow FS doesn?t load it. What could be the problem? Thanks _____ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/fd371697/attachment.html From anthony.minessale at gmail.com Wed May 13 05:58:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 13 May 2009 07:58:51 -0500 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <1242199827.10932.1.camel@mikael-xpsm1530> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> <23516585.post@talk.nabble.com> <1242199827.10932.1.camel@mikael-xpsm1530> Message-ID: <191c3a030905130558j710076c9h2f5530f3c87555cd@mail.gmail.com> you *need* park because you have to have somewhere to anchor the call to and the park function is the routine that actually parses all the DTMF and the commands you send the channel with sendmsg etc. I would have to look into refactoring it so when there is no media on the channel, it would sleep in place of reading audio but if you send it any instruction that required media like playback etc it would still instantly send a 183. Don't forget we are a b2bua here so proxying calls is only smoke and mirrors for us. Just as if you put a playback "please_wait.wav" before bridge, when you try to use a media enabled app it will automatically generate a 183 to establish early media to make it possible. I'll see what I can do, it may be difficult to avoid regressions, don't forget my wishlist if i pull it off ;) On Wed, May 13, 2009 at 2:30 AM, Mikael Aleksander Bjerkeland < mikael at bjerkeland.com> wrote: > I've been looking for something like this as well, to make the outbound > event socket behave more like FastAGI and handle the logic. > > > El mi?, 13-05-2009 a las 00:12 -0700, ibrahim tunali escribi?: > > "nopark" option would be great. FS sends "100 trying" while opening > socket > > and leg A waits other responses from socket. If I send pre-answer command > it > > reply via 183 early media and activate RTP path. > > > > Regards, > > ibrahim > > > > > > David Knell wrote: > > > > > > Gotcha - but in the case where the call hasn't yet got to a point where > > > there's a 183 been sent then I guess this wouldn't apply - there > > > shouldn't be any audio from the far end at this point, nor would the > far > > > end be expecting any. > > > > > > I'd suggest (and would volunteer to knock together a patch) adding a > > > 'nopark' option to the socket command, which doesn't park the call - > nor > > > would it change existing behaviour. Obviously, in a situation like > that > > > outlined by Ibrahim where the socket app handles all aspects of the > > > call, then it'll need to make sure that it signals ringing, answer or > > > whatever to make the call state flow work. > > > > > > --Dave > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/b6c85433/attachment.html From mikael at bjerkeland.com Wed May 13 06:15:49 2009 From: mikael at bjerkeland.com (Mikael Aleksander Bjerkeland) Date: Wed, 13 May 2009 15:15:49 +0200 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <191c3a030905130558j710076c9h2f5530f3c87555cd@mail.gmail.com> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> <23516585.post@talk.nabble.com> <1242199827.10932.1.camel@mikael-xpsm1530> <191c3a030905130558j710076c9h2f5530f3c87555cd@mail.gmail.com> Message-ID: <1242220549.3935.4.camel@mikael-xpsm1530> If you could make it in such a way that 183 or early media is sent only when media is really needed like you suggested yourself, it would work in the exact way most users would expect. A clarification of what happens if I do transfer or execute_extension on a socket parked call would also be neat. I remember reading that a transfer would alter the state of the call, but I haven't seen any proper documentation for this claim. Maybe if you're lucky Santa comes earlier this year. El mi?, 13-05-2009 a las 07:58 -0500, Anthony Minessale escribi?: > you *need* park because you have to have somewhere to anchor the call > to and the park function is the routine > that actually parses all the DTMF and the commands you send the > channel with sendmsg etc. > > I would have to look into refactoring it so when there is no media on > the channel, it would sleep in place > of reading audio but if you send it any instruction that required > media like playback etc it would still instantly send > a 183. > > Don't forget we are a b2bua here so proxying calls is only smoke and > mirrors for us. > > Just as if you put a playback "please_wait.wav" before bridge, when > you try to use a media enabled app > it will automatically generate a 183 to establish early media to make > it possible. > > I'll see what I can do, it may be difficult to avoid regressions, > don't forget my wishlist if i pull it off ;) > > > > > On Wed, May 13, 2009 at 2:30 AM, Mikael Aleksander Bjerkeland > wrote: > I've been looking for something like this as well, to make the > outbound > event socket behave more like FastAGI and handle the logic. > > > El mi?, 13-05-2009 a las 00:12 -0700, ibrahim tunali escribi?: > > "nopark" option would be great. FS sends "100 trying" while > opening socket > > and leg A waits other responses from socket. If I send > pre-answer command it > > reply via 183 early media and activate RTP path. > > > > Regards, > > ibrahim > > > > > > David Knell wrote: > > > > > > Gotcha - but in the case where the call hasn't yet got to > a point where > > > there's a 183 been sent then I guess this wouldn't apply - > there > > > shouldn't be any audio from the far end at this point, nor > would the far > > > end be expecting any. > > > > > > I'd suggest (and would volunteer to knock together a > patch) adding a > > > 'nopark' option to the socket command, which doesn't park > the call - nor > > > would it change existing behaviour. Obviously, in a > situation like that > > > outlined by Ibrahim where the socket app handles all > aspects of the > > > call, then it'll need to make sure that it signals > ringing, answer or > > > whatever to make the call state flow work. > > > > > > --Dave > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vhatz at kinetix.gr Wed May 13 06:26:10 2009 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Wed, 13 May 2009 16:26:10 +0300 Subject: [Freeswitch-users] Disable early media (how to)? Message-ID: <4A0ACA72.9000509@kinetix.gr> Hello, I am trying to understand how to disable early media in both directions during the call negotiation. In the following scenario: Originator SIP Client -- SIP --> FS with ignore_early_media=true, FS starts sending RTP audio to the originator right after the "200 OK" is sent from FS to originator. This is correct. However, FS sends SDP information in the Session Progress (183) message before the "200 OK", and the originator client sends RTP before the call is answered. The strange thing is that SDP is sent again in the "200" message although it is already sent in the 183 message (I don't know if this redundant behavior is acceptable in SIP, I have a mostly H323 background) So, in this way early media is disabled only for the audio from FS to the client, but not from the client to FS. My question is: how can I prevent FS from sending SDP in the 183 message and send it only in the 200 message? I searched the mailing list archives but didn't come up with anything. Any help is appreciated. -- Best regards, Vlasis Hatzistavrou. From rupa at rupa.com Wed May 13 06:46:37 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 13 May 2009 08:46:37 -0500 Subject: [Freeswitch-users] Set nibblebill variables on B-leg In-Reply-To: <200905131332.35250.yivzhenko@mksat.net> References: <200905131332.35250.yivzhenko@mksat.net> Message-ID: Put them on the dialstring: application="bridge" data="{nibblevar1=val1,nibblevar2=val2}sofia/blah...." On Wed, May 13, 2009 at 5:32 AM, Yuriy Ivzhenko wrote: > Hi All, > > I want to use mod_nibblebill for billing outgoing calls. > > My problem is billing begins before other paty answers the call. > > The WiKi tells: "You need to set the billing variables on your outbound > calling leg and NOT on your A-Leg." > > How i can set nibblebill variables when B-leg connects? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/86e82f36/attachment.html From brian at freeswitch.org Wed May 13 07:08:22 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 09:08:22 -0500 Subject: [Freeswitch-users] Set nibblebill variables on B-leg In-Reply-To: References: <200905131332.35250.yivzhenko@mksat.net> Message-ID: <6FD41C9B-DBC2-4C59-8B18-3822941ABF00@freeswitch.org> or use export vs set. /b On May 13, 2009, at 8:46 AM, Rupa Schomaker wrote: > Put them on the dialstring: > > application="bridge" data="{nibblevar1=val1,nibblevar2=val2}sofia/ > blah...." Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/7e9ec640/attachment.html From helmut.kuper at ewetel.de Wed May 13 07:26:41 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 13 May 2009 16:26:41 +0200 Subject: [Freeswitch-users] Question about ACL Message-ID: <4A0AD8A1.9060907@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I found tha FS isn't authenticating incomming calls from local extensions. I use an ACL and I have auth-calls set to true in default sip profile. FS log shows, that ACL is granting access but FS doesn't authenticate the call. When I disable the ACL in default profile, authentication is performed. FS WIKI doku (http://wiki.freeswitch.org/wiki/Acl#Sofia) chapter "sip_profiles" says this: "The acl behaviour is modfied by auth-calls, accept-blind-reg and accept-blind-auth" I rather think ACL disables those parameters. What I want is an ACL check followed by authentication. I use "FreeSWITCH Version 1.0.trunk (13112M)" Any ideas? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCtih4tZeNddg3dwRAjtYAJsGqPjBp+FVEKZl4GTQNq3TLoh+JwCgiUiZ wJcqu2XOQ7TV+0tBU8REEFc= =M/pp -----END PGP SIGNATURE----- From brian at freeswitch.org Wed May 13 07:32:22 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 09:32:22 -0500 Subject: [Freeswitch-users] Question about ACL In-Reply-To: <4A0AD8A1.9060907@ewetel.de> References: <4A0AD8A1.9060907@ewetel.de> Message-ID: <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> Wouldn't disable be a modification? Anyway the domains ACL from acl.conf.xml reads all the cidr= tags in the domains specified to build the ACL /b On May 13, 2009, at 9:26 AM, Helmut Kuper wrote: > Hash: SHA1 > > Hello, > > today I found tha FS isn't authenticating incomming calls from local > extensions. I use an ACL and I have auth-calls set to true in default > sip profile. > > > > > FS log shows, that ACL is granting access but FS doesn't authenticate > the call. When I disable the ACL in default profile, authentication is > performed. > > FS WIKI doku (http://wiki.freeswitch.org/wiki/Acl#Sofia) chapter > "sip_profiles" says this: > > "The acl behaviour is modfied by auth-calls, accept-blind-reg and > accept-blind-auth" > > I rather think ACL disables those parameters. > > What I want is an ACL check followed by authentication. > > I use "FreeSWITCH Version 1.0.trunk (13112M)" > > Any ideas? > > regards > helmut Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/80562ba9/attachment.html From helmut.kuper at ewetel.de Wed May 13 07:50:55 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 13 May 2009 16:50:55 +0200 Subject: [Freeswitch-users] Question about ACL In-Reply-To: <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> References: <4A0AD8A1.9060907@ewetel.de> <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> Message-ID: <4A0ADE4F.1090607@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Brian, yes indeed, disabling is a kind of modification. Hm, I' not sure if we misunderstand. ACL is working. auth-calls also. But together ACL disables somehow authentication. I'm not using cidr in my directory. Is there a way to get both working like this: 1. Calls comes in 2. ACL is applied 3. Authentication is performed 4. FS is executing the call regards Helmut On 13.05.2009 16:32, Brian West wrote: > Wouldn't disable be a modification? Anyway the domains ACL from > acl.conf.xml reads all the cidr= tags in the domains specified to build > the ACL > > /b > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCt5P4tZeNddg3dwRAqGWAJ4owWbow3QWNSlM6y9u2VuLTe9UVQCfZkq6 HkwAX2o7Rd0G+ElsVUMuy10= =mDEj -----END PGP SIGNATURE----- From mail-lists at peachnet.com Wed May 13 08:02:31 2009 From: mail-lists at peachnet.com (mail-lists) Date: Wed, 13 May 2009 11:02:31 -0400 Subject: [Freeswitch-users] Fax through FS to Callweaver. How? In-Reply-To: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> References: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> Message-ID: <4A0AE107.5000202@peachnet.com> Dennis, I've been playing with fs/callweaver for a couple weeks. I have fs routing calls and passing them to callweaver. In proxy_media mode it works perfectly in bypass_media mode it works only sometimes. Are you doing an 'answer' on FS before you bridge the call? Are you sure that the call is being put in proxy_media mode? Do you have canreinvite=yes on callweaver set? I also had to put the following in my extensions.conf: exten => s,n,Set(SIP_CODEC=ulaw) exten => s,n,SipT38SwitchOver() Those are some of the things I played with to get it working. Let me know how you make out. Fax over IP is an odyssey. > hi, > > we are doing a lot with freeswitch and everything works absolutely perfect. > > the only thing we are missing, is a good support for fax over ip - may > it be for offering a fax-2-mail service or just passing incoming > fax-calls from our fs-servers to a "real" fax-machine. > > now we have another server only for callweaver, which has a great > fax-support. if we send faxes from our fax-machine over our > sip-provider (who supports t.38) directly to the cw-server, all faxes > are beeing received and saved by cw. but if we send faxes over the > same sip-provider first to fs and then do a bridge to the cw-server, > all faxes fail. > > we set in > internal.xml and play with bypass_media=true and/or proxy_media=true > in the dialplan. it simply does not work. > > i think and hope that there are some other settings i can change, so > that fs is nothing more than just a "tunnel" and let both sides > communicate with eachother? > > does someone know callweaver and can tell me, if there are some > important settings to be set for making it work with fs in the middle? > > > thanks & kind regards > dennis > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed May 13 08:03:24 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 10:03:24 -0500 Subject: [Freeswitch-users] Question about ACL In-Reply-To: <4A0ADE4F.1090607@ewetel.de> References: <4A0AD8A1.9060907@ewetel.de> <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> <4A0ADE4F.1090607@ewetel.de> Message-ID: <90D8063F-145D-40D0-BD7F-CB0CE9D08E43@freeswitch.org> On May 13, 2009, at 9:50 AM, Helmut Kuper wrote: > Hi Brian, > > yes indeed, disabling is a kind of modification. > > > Hm, I' not sure if we misunderstand. ACL is working. auth-calls also. > But together ACL disables somehow authentication. I'm not using cidr > in > my directory. That means you do have somewhere in your ACL the IP's, I would double check that. Did you happen to change your ACL to default allow? Show me your acl.conf.xml > > Is there a way to get both working like this: > > 1. Calls comes in > 2. ACL is applied > 3. Authentication is performed > 4. FS is executing the call > > regards > Helmut Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/d9a12414/attachment-0001.html From gallo at mctelefonia.com Wed May 13 08:23:44 2009 From: gallo at mctelefonia.com (Antonio Gallo) Date: Wed, 13 May 2009 17:23:44 +0200 Subject: [Freeswitch-users] Fax through FS to Callweaver. How? In-Reply-To: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> References: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> Message-ID: <4A0AE600.3040401@mctelefonia.com> Dennis ha scritto: > does someone know callweaver and can tell me, if there are some > important settings to be set for making it work with fs in the middle? > Look at this, i needed to apply it using a Patton gateway too: http://www.callweaver.org/ticket/487 From helmut.kuper at ewetel.de Wed May 13 08:30:53 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 13 May 2009 17:30:53 +0200 Subject: [Freeswitch-users] Question about ACL In-Reply-To: <90D8063F-145D-40D0-BD7F-CB0CE9D08E43@freeswitch.org> References: <4A0AD8A1.9060907@ewetel.de> <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> <4A0ADE4F.1090607@ewetel.de> <90D8063F-145D-40D0-BD7F-CB0CE9D08E43@freeswitch.org> Message-ID: <4A0AE7AD.3050001@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Brian, yes, you are right, I defined the phone network as an ACL and I want it. Obvoiusly I didn't explained my problem clearly. I want both: ACL *AND* SIP-Authentication vor each call and each local extension/user, but I'm only able to have ACL *OR* SIP-Auth. Is there a way to configure it? regards Helmut On 13.05.2009 17:03, Brian West wrote: > > On May 13, 2009, at 9:50 AM, Helmut Kuper wrote: > >> Hi Brian, >> >> yes indeed, disabling is a kind of modification. >> >> >> Hm, I' not sure if we misunderstand. ACL is working. auth-calls also. >> But together ACL disables somehow authentication. I'm not using cidr in >> my directory. > > > That means you do have somewhere in your ACL the IP's, I would double > check that. > > Did you happen to change your ACL to default allow? > > Show me your acl.conf.xml > > >> >> Is there a way to get both working like this: >> >> 1. Calls comes in >> 2. ACL is applied >> 3. Authentication is performed >> 4. FS is executing the call >> >> regards >> Helmut > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org - -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCuet4tZeNddg3dwRAloYAJ9XHbybGqLQWJ1xN+d0i04BK8D/8QCbBHHv 7zWauA2VyTC1lzJJE8nH8zc= =Atsf -----END PGP SIGNATURE----- From helmut.kuper at ewetel.de Wed May 13 08:33:10 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 13 May 2009 17:33:10 +0200 Subject: [Freeswitch-users] RFC: pickup using a two nodes FS PBX In-Reply-To: <4A0A7CF8.7040809@ewetel.de> References: <4A0A7CF8.7040809@ewetel.de> Message-ID: <4A0AE836.4070108@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello all, I found a solution by myself. A modified mod_limit and FS sip gateway functionality was the answer. Works fine at least in in Lab :) regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCug24tZeNddg3dwRAi8kAJ9dOCLDB2Bgl5ljINiAO4gg4VMegQCeLBCZ XZH/pxY9+eItqQyJMgOblAY= =s7Bo -----END PGP SIGNATURE----- From brian at freeswitch.org Wed May 13 08:34:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 10:34:42 -0500 Subject: [Freeswitch-users] Question about ACL In-Reply-To: <4A0AE7AD.3050001@ewetel.de> References: <4A0AD8A1.9060907@ewetel.de> <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> <4A0ADE4F.1090607@ewetel.de> <90D8063F-145D-40D0-BD7F-CB0CE9D08E43@freeswitch.org> <4A0AE7AD.3050001@ewetel.de> Message-ID: <7B10AAB2-DC37-4A50-AAC8-97A132F96468@freeswitch.org> No you can't have ACL AND SIP auth at the same time.. ACL lets them in... you can however use respond 407 in the dialplan to cause it to auth again. /b On May 13, 2009, at 10:30 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello Brian, > > > yes, you are right, I defined the phone network as an ACL and I want > it. > Obvoiusly I didn't explained my problem clearly. > > I want both: ACL *AND* SIP-Authentication vor each call and each local > extension/user, but I'm only able to have ACL *OR* SIP-Auth. > > > Is there a way to configure it? > > regards > Helmut Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/927e1abc/attachment.html From red.rain.seven at gmail.com Wed May 13 09:13:20 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 13 May 2009 09:13:20 -0700 Subject: [Freeswitch-users] RFC: pickup using a two nodes FS PBX In-Reply-To: <4A0AE836.4070108@ewetel.de> References: <4A0A7CF8.7040809@ewetel.de> <4A0AE836.4070108@ewetel.de> Message-ID: <59ad9ca10905130913n60dc04c3g38d9a2ec04c4d1fd@mail.gmail.com> Hi: Can you share it in the wiki please? On Wed, May 13, 2009 at 8:33 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello all, > > I found a solution by myself. A modified mod_limit and FS sip gateway > functionality was the answer. Works fine at least in in Lab :) > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKCug24tZeNddg3dwRAi8kAJ9dOCLDB2Bgl5ljINiAO4gg4VMegQCeLBCZ > XZH/pxY9+eItqQyJMgOblAY= > =s7Bo > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/07dc40b2/attachment.html From brian at freeswitch.org Wed May 13 09:36:10 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 11:36:10 -0500 Subject: [Freeswitch-users] Question about ACL In-Reply-To: <4A0AE7AD.3050001@ewetel.de> References: <4A0AD8A1.9060907@ewetel.de> <1D81621A-1F11-494C-B57D-76EDF2009996@freeswitch.org> <4A0ADE4F.1090607@ewetel.de> <90D8063F-145D-40D0-BD7F-CB0CE9D08E43@freeswitch.org> <4A0AE7AD.3050001@ewetel.de> Message-ID: <607D4B92-32C5-48E7-8DCA-6F0AEBD40B1C@freeswitch.org> Don't use the ACL on the profile and use the ACL command in the dialplan then you can do it. /b On May 13, 2009, at 10:30 AM, Helmut Kuper wrote: > I want both: ACL *AND* SIP-Authentication vor each call and each local > extension/user, but I'm only able to have ACL *OR* SIP-Auth. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/dcaac180/attachment.html From anthony.minessale at gmail.com Wed May 13 09:38:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 13 May 2009 11:38:00 -0500 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <1242220549.3935.4.camel@mikael-xpsm1530> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> <23516585.post@talk.nabble.com> <1242199827.10932.1.camel@mikael-xpsm1530> <191c3a030905130558j710076c9h2f5530f3c87555cd@mail.gmail.com> <1242220549.3935.4.camel@mikael-xpsm1530> Message-ID: <191c3a030905130938x50b6e3b7q36ded153d5110f6f@mail.gmail.com> please test r13294 ASAP i do not want to create any regressions that last more than a day. On Wed, May 13, 2009 at 8:15 AM, Mikael Aleksander Bjerkeland < mikael at bjerkeland.com> wrote: > If you could make it in such a way that 183 or early media is sent only > when media is really needed like you suggested yourself, it would work > in the exact way most users would expect. > > A clarification of what happens if I do transfer or execute_extension on > a socket parked call would also be neat. I remember reading that a > transfer would alter the state of the call, but I haven't seen any > proper documentation for this claim. > > Maybe if you're lucky Santa comes earlier this year. > > > El mi?, 13-05-2009 a las 07:58 -0500, Anthony Minessale escribi?: > > you *need* park because you have to have somewhere to anchor the call > > to and the park function is the routine > > that actually parses all the DTMF and the commands you send the > > channel with sendmsg etc. > > > > I would have to look into refactoring it so when there is no media on > > the channel, it would sleep in place > > of reading audio but if you send it any instruction that required > > media like playback etc it would still instantly send > > a 183. > > > > Don't forget we are a b2bua here so proxying calls is only smoke and > > mirrors for us. > > > > Just as if you put a playback "please_wait.wav" before bridge, when > > you try to use a media enabled app > > it will automatically generate a 183 to establish early media to make > > it possible. > > > > I'll see what I can do, it may be difficult to avoid regressions, > > don't forget my wishlist if i pull it off ;) > > > > > > > > > > On Wed, May 13, 2009 at 2:30 AM, Mikael Aleksander Bjerkeland > > wrote: > > I've been looking for something like this as well, to make the > > outbound > > event socket behave more like FastAGI and handle the logic. > > > > > > El mi?, 13-05-2009 a las 00:12 -0700, ibrahim tunali escribi?: > > > "nopark" option would be great. FS sends "100 trying" while > > opening socket > > > and leg A waits other responses from socket. If I send > > pre-answer command it > > > reply via 183 early media and activate RTP path. > > > > > > Regards, > > > ibrahim > > > > > > > > > David Knell wrote: > > > > > > > > Gotcha - but in the case where the call hasn't yet got to > > a point where > > > > there's a 183 been sent then I guess this wouldn't apply - > > there > > > > shouldn't be any audio from the far end at this point, nor > > would the far > > > > end be expecting any. > > > > > > > > I'd suggest (and would volunteer to knock together a > > patch) adding a > > > > 'nopark' option to the socket command, which doesn't park > > the call - nor > > > > would it change existing behaviour. Obviously, in a > > situation like that > > > > outlined by Ibrahim where the socket app handles all > > aspects of the > > > > call, then it'll need to make sure that it signals > > ringing, answer or > > > > whatever to make the call state flow work. > > > > > > > > --Dave > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/4e257ca9/attachment-0001.html From mike at jerris.com Wed May 13 10:11:37 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 13 May 2009 13:11:37 -0400 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <2F7A3C109BB644DAAD92816534F1948F@saeedlaptop> References: <2059908291DA4436B2009BBB52584C46@saeedlaptop> <4A0AB9CF.4030307@telemaque.fr> <2F7A3C109BB644DAAD92816534F1948F@saeedlaptop> Message-ID: do you see any errors if you do a make ? Mike On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote: > Hi Tristan, > > No its not commented out. > > And strange thing is that it was working until I did ?make current? > today. > > Thanks for your response. > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Tristan > Sent: Wednesday, May 13, 2009 2:15 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk > 13288 > > Hi Saeed, > > Maybe you did not built it ? > > ( see in modules.conf if it's not commented out with a # ) > > Saeed Ahmed a ?crit : > Hi, > > I just did ?make current? and FS failed to load xml_curl, I looked > into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy > it from src directory but somehow FS doesn?t load it. What could be > the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/e2c09fc7/attachment.html From mikael at bjerkeland.com Wed May 13 13:30:48 2009 From: mikael at bjerkeland.com (Mikael Bjerkeland) Date: Wed, 13 May 2009 22:30:48 +0200 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: <191c3a030905130938x50b6e3b7q36ded153d5110f6f@mail.gmail.com> References: <191c3a030905120800x4a89401ap3e30f0d30efa46c7@mail.gmail.com> <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> <23516585.post@talk.nabble.com> <1242199827.10932.1.camel@mikael-xpsm1530> <191c3a030905130558j710076c9h2f5530f3c87555cd@mail.gmail.com> <1242220549.3935.4.camel@mikael-xpsm1530> <191c3a030905130938x50b6e3b7q36ded153d5110f6f@mail.gmail.com> Message-ID: Connection to "Action socket(localhost:8086 full)" works! No early media there. Scenarios tested: * Just setting a few vars, then bridge attempt to a phone generates 180 ringing - as expected * Answer, then playback answers the call and plays audio - as expected * Playback sends 183 session progress and plays audio - as expected Great work. No regressions seen in my scenario. 2009/5/13 Anthony Minessale > please test r13294 ASAP > i do not want to create any regressions that last more than a day. > > > > > > On Wed, May 13, 2009 at 8:15 AM, Mikael Aleksander Bjerkeland < > mikael at bjerkeland.com> wrote: > >> If you could make it in such a way that 183 or early media is sent only >> when media is really needed like you suggested yourself, it would work >> in the exact way most users would expect. >> >> A clarification of what happens if I do transfer or execute_extension on >> a socket parked call would also be neat. I remember reading that a >> transfer would alter the state of the call, but I haven't seen any >> proper documentation for this claim. >> >> Maybe if you're lucky Santa comes earlier this year. >> >> >> El mi?, 13-05-2009 a las 07:58 -0500, Anthony Minessale escribi?: >> > you *need* park because you have to have somewhere to anchor the call >> > to and the park function is the routine >> > that actually parses all the DTMF and the commands you send the >> > channel with sendmsg etc. >> > >> > I would have to look into refactoring it so when there is no media on >> > the channel, it would sleep in place >> > of reading audio but if you send it any instruction that required >> > media like playback etc it would still instantly send >> > a 183. >> > >> > Don't forget we are a b2bua here so proxying calls is only smoke and >> > mirrors for us. >> > >> > Just as if you put a playback "please_wait.wav" before bridge, when >> > you try to use a media enabled app >> > it will automatically generate a 183 to establish early media to make >> > it possible. >> > >> > I'll see what I can do, it may be difficult to avoid regressions, >> > don't forget my wishlist if i pull it off ;) >> > >> > >> > >> > >> > On Wed, May 13, 2009 at 2:30 AM, Mikael Aleksander Bjerkeland >> > wrote: >> > I've been looking for something like this as well, to make the >> > outbound >> > event socket behave more like FastAGI and handle the logic. >> > >> > >> > El mi?, 13-05-2009 a las 00:12 -0700, ibrahim tunali escribi?: >> > > "nopark" option would be great. FS sends "100 trying" while >> > opening socket >> > > and leg A waits other responses from socket. If I send >> > pre-answer command it >> > > reply via 183 early media and activate RTP path. >> > > >> > > Regards, >> > > ibrahim >> > > >> > > >> > > David Knell wrote: >> > > > >> > > > Gotcha - but in the case where the call hasn't yet got to >> > a point where >> > > > there's a 183 been sent then I guess this wouldn't apply - >> > there >> > > > shouldn't be any audio from the far end at this point, nor >> > would the far >> > > > end be expecting any. >> > > > >> > > > I'd suggest (and would volunteer to knock together a >> > patch) adding a >> > > > 'nopark' option to the socket command, which doesn't park >> > the call - nor >> > > > would it change existing behaviour. Obviously, in a >> > situation like that >> > > > outlined by Ibrahim where the socket app handles all >> > aspects of the >> > > > call, then it'll need to make sure that it signals >> > ringing, answer or >> > > > whatever to make the call state flow work. >> > > > >> > > > --Dave >> > > > >> > > > >> > > >> > >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > iax:guest at conference.freeswitch.org/888 >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:213-799-1400 >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/f2372b7f/attachment.html From msc at freeswitch.org Wed May 13 15:56:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 May 2009 15:56:16 -0700 Subject: [Freeswitch-users] mod_event_socket 183 early media In-Reply-To: References: <1242150564.4019.6.camel@dk-d820> <191c3a030905121113v3c022799ic7ee5ff500d6aa80@mail.gmail.com> <1242170592.4019.13.camel@dk-d820> <23516585.post@talk.nabble.com> <1242199827.10932.1.camel@mikael-xpsm1530> <191c3a030905130558j710076c9h2f5530f3c87555cd@mail.gmail.com> <1242220549.3935.4.camel@mikael-xpsm1530> <191c3a030905130938x50b6e3b7q36ded153d5110f6f@mail.gmail.com> Message-ID: <87f2f3b90905131556t73a62dcbu1e2597f600979428@mail.gmail.com> Anthm's paypal link is conveniently located on the right side of the main FreeSWITCH page... :) -MC On Wed, May 13, 2009 at 1:30 PM, Mikael Bjerkeland wrote: > Connection to "Action socket(localhost:8086 full)" works! No early media > there. Scenarios tested: > > * Just setting a few vars, then bridge attempt to a phone generates 180 > ringing - as expected > * Answer, then playback answers the call and plays audio - as expected > * Playback sends 183 session progress and plays audio - as expected > > > Great work. No regressions seen in my scenario. > > > > 2009/5/13 Anthony Minessale > >> please test r13294 ASAP >> i do not want to create any regressions that last more than a day. >> >> >> >> >> >> On Wed, May 13, 2009 at 8:15 AM, Mikael Aleksander Bjerkeland < >> mikael at bjerkeland.com> wrote: >> >>> If you could make it in such a way that 183 or early media is sent only >>> when media is really needed like you suggested yourself, it would work >>> in the exact way most users would expect. >>> >>> A clarification of what happens if I do transfer or execute_extension on >>> a socket parked call would also be neat. I remember reading that a >>> transfer would alter the state of the call, but I haven't seen any >>> proper documentation for this claim. >>> >>> Maybe if you're lucky Santa comes earlier this year. >>> >>> >>> El mi?, 13-05-2009 a las 07:58 -0500, Anthony Minessale escribi?: >>> > you *need* park because you have to have somewhere to anchor the call >>> > to and the park function is the routine >>> > that actually parses all the DTMF and the commands you send the >>> > channel with sendmsg etc. >>> > >>> > I would have to look into refactoring it so when there is no media on >>> > the channel, it would sleep in place >>> > of reading audio but if you send it any instruction that required >>> > media like playback etc it would still instantly send >>> > a 183. >>> > >>> > Don't forget we are a b2bua here so proxying calls is only smoke and >>> > mirrors for us. >>> > >>> > Just as if you put a playback "please_wait.wav" before bridge, when >>> > you try to use a media enabled app >>> > it will automatically generate a 183 to establish early media to make >>> > it possible. >>> > >>> > I'll see what I can do, it may be difficult to avoid regressions, >>> > don't forget my wishlist if i pull it off ;) >>> > >>> > >>> > >>> > >>> > On Wed, May 13, 2009 at 2:30 AM, Mikael Aleksander Bjerkeland >>> > wrote: >>> > I've been looking for something like this as well, to make the >>> > outbound >>> > event socket behave more like FastAGI and handle the logic. >>> > >>> > >>> > El mi?, 13-05-2009 a las 00:12 -0700, ibrahim tunali escribi?: >>> > > "nopark" option would be great. FS sends "100 trying" while >>> > opening socket >>> > > and leg A waits other responses from socket. If I send >>> > pre-answer command it >>> > > reply via 183 early media and activate RTP path. >>> > > >>> > > Regards, >>> > > ibrahim >>> > > >>> > > >>> > > David Knell wrote: >>> > > > >>> > > > Gotcha - but in the case where the call hasn't yet got to >>> > a point where >>> > > > there's a 183 been sent then I guess this wouldn't apply - >>> > there >>> > > > shouldn't be any audio from the far end at this point, nor >>> > would the far >>> > > > end be expecting any. >>> > > > >>> > > > I'd suggest (and would volunteer to knock together a >>> > patch) adding a >>> > > > 'nopark' option to the socket command, which doesn't park >>> > the call - nor >>> > > > would it change existing behaviour. Obviously, in a >>> > situation like that >>> > > > outlined by Ibrahim where the socket app handles all >>> > aspects of the >>> > > > call, then it'll need to make sure that it signals >>> > ringing, answer or >>> > > > whatever to make the call state flow work. >>> > > > >>> > > > --Dave >>> > > > >>> > > > >>> > > >>> > >>> > >>> > >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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 at hotmail.com >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> > iax:guest at conference.freeswitch.org/888 >>> > googletalk:conf+888 at conference.freeswitch.org >>> > pstn:213-799-1400 >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/e29449d6/attachment-0001.html From brian at freeswitch.org Wed May 13 17:27:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 19:27:21 -0500 Subject: [Freeswitch-users] Polycom and SRTP Message-ID: <5A2D6DB9-61D0-4498-8223-A0C5F79D0DB1@freeswitch.org> I finally narrowed down what the heck the polycom wanted when the hold operation was taking place. Now Please test snom, polycom and others that use SRTP. I have tested SNOM and Polycom but others that use SRTP will need to test it also. All I can say is DUMB DUMB DUMB DUMB DUMB!!!!! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/b6468ba5/attachment.html From jason at jasonjgw.net Wed May 13 19:50:53 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 14 May 2009 12:50:53 +1000 Subject: [Freeswitch-users] Suppressing music on hold in conferences Message-ID: <20090514025053.GA30945@jdc.jasonjgw.net> >From the default dialplan: I have just written a similar entry for another conference that I frequently call. If I call the conference from FreeSWITCH via a SIP phone, then place the call on hold, the hold music is suppressed as intended. However, if I call the conference from PortAudio, the above dial-plan logic doesn't have the desired effect - I still get music on hold. The info application shows that the variable is being exported, however. This is under revision 13266. Is this a bug, or is there additional dialplan logic that I need to supply? If it's a bug, could someone add it to the bug list? Thanks. From brian at freeswitch.org Wed May 13 20:09:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 22:09:44 -0500 Subject: [Freeswitch-users] Suppressing music on hold in conferences In-Reply-To: <20090514025053.GA30945@jdc.jasonjgw.net> References: <20090514025053.GA30945@jdc.jasonjgw.net> Message-ID: <7B9C0056-75D7-42AD-89F0-A7A92400EAA9@freeswitch.org> Confirmed bug... I'll see if I can get that fixed rather quickly. /b On May 13, 2009, at 9:50 PM, Jason White wrote: > > application="export" data="hold_music=silence"/> > > I have just written a similar entry for another conference that I > frequently > call. If I call the conference from FreeSWITCH via a SIP phone, then > place the > call on hold, the hold music is suppressed as intended. However, if > I call the > conference from PortAudio, the above dial-plan logic doesn't have > the desired > effect - I still get music on hold. The info application shows that > the > variable is being exported, however. > > This is under revision 13266. > > Is this a bug, or is there additional dialplan logic that I need to > supply? > > If it's a bug, could someone add it to the bug list? > > Thanks. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/4509734a/attachment.html From brian at freeswitch.org Wed May 13 20:18:24 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 13 May 2009 22:18:24 -0500 Subject: [Freeswitch-users] Suppressing music on hold in conferences In-Reply-To: <20090514025053.GA30945@jdc.jasonjgw.net> References: <20090514025053.GA30945@jdc.jasonjgw.net> Message-ID: It seems to be the global hold_music var is not being overridden /b On May 13, 2009, at 9:50 PM, Jason White wrote: > > application="export" data="hold_music=silence"/> > > I have just written a similar entry for another conference that I > frequently > call. If I call the conference from FreeSWITCH via a SIP phone, then > place the > call on hold, the hold music is suppressed as intended. However, if > I call the > conference from PortAudio, the above dial-plan logic doesn't have > the desired > effect - I still get music on hold. The info application shows that > the > variable is being exported, however. > > This is under revision 13266. > > Is this a bug, or is there additional dialplan logic that I need to > supply? > > If it's a bug, could someone add it to the bug list? > > Thanks. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090513/017e86b1/attachment.html From yivzhenko at mksat.net Thu May 14 00:19:01 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 14 May 2009 10:19:01 +0300 Subject: [Freeswitch-users] Set nibblebill variables on B-leg In-Reply-To: References: <200905131332.35250.yivzhenko@mksat.net> Message-ID: <200905141019.02081.yivzhenko@mksat.net> Thanks for answer, i try to but it not resolve my problem. Billing still begins on A-leg before the B-leg answered. So, bill time include ringback time. Here is some debug output: EXECUTE ............... info() ... Unique-ID: [9441ad16-3fce-11de-baaf-f74f6eef9491] ... ... [DEBUG] switch_ivr_originate.c:1708 switch_ivr_originate() Play Ringback File [local_stream://rb_music1] [DEBUG] mod_local_stream.c:346 local_stream_file_open() Opening Stream [rb_music1] 8000hz [DEBUG] mod_nibblebill.c:459 event_handler() Received request via SESSION_HEARTBEAT! [DEBUG] mod_nibblebill.c:345 do_billing() Attempting to bill at $6.00000 per minute to account 3 [INFO] mod_nibblebill.c:387 do_billing() Beginning new billing on 9441ad16-3fce-11de-baaf-f74f6eef9491 [DEBUG] mod_nibblebill.c:395 do_billing() 0 seconds passed since last bill time of 2009-05-13 17:58:55 [DEBUG] mod_nibblebill.c:402 do_billing() Billing $0.036643 to 3 (Call: 9441ad16-3fce-11de-baaf-f74f6eef9491 / 0.000000 so far) On Wednesday 13 May 2009 16:46:37 Rupa Schomaker wrote: > Put them on the dialstring: > > application="bridge" data="{nibblevar1=val1,nibblevar2=val2}sofia/blah...." > > On Wed, May 13, 2009 at 5:32 AM, Yuriy Ivzhenko wrote: > > Hi All, > > > > I want to use mod_nibblebill for billing outgoing calls. > > > > My problem is billing begins before other paty answers the call. > > > > The WiKi tells: "You need to set the billing variables on your outbound > > calling leg and NOT on your A-Leg." > > > > How i can set nibblebill variables when B-leg connects? > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/0fcc2fe1/attachment-0001.html From krice at freeswitch.org Thu May 14 01:07:19 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 14 May 2009 03:07:19 -0500 Subject: [Freeswitch-users] Set nibblebill variables on B-leg In-Reply-To: <200905141019.02081.yivzhenko@mksat.net> Message-ID: That?s because on your bridge line if that?s exactly what you have in your dialplan its wrong... You don?t use square brackets but the curly braces... {variable=123} From: Yuriy Ivzhenko Organization: WildPark Reply-To: Date: Thu, 14 May 2009 10:19:01 +0300 To: Subject: Re: [Freeswitch-users] Set nibblebill variables on B-leg Thanks for answer, i try to but it not resolve my problem. Billing still begins on A-leg before the B-leg answered. So, bill time include ringback time. Here is some debug output: EXECUTE ............... info() ... Unique-ID: [9441ad16-3fce-11de-baaf-f74f6eef9491] ... ... [DEBUG] switch_ivr_originate.c:1708 switch_ivr_originate() Play Ringback File [local_stream://rb_music1] [DEBUG] mod_local_stream.c:346 local_stream_file_open() Opening Stream [rb_music1] 8000hz [DEBUG] mod_nibblebill.c:459 event_handler() Received request via SESSION_HEARTBEAT! [DEBUG] mod_nibblebill.c:345 do_billing() Attempting to bill at $6.00000 per minute to account 3 [INFO] mod_nibblebill.c:387 do_billing() Beginning new billing on 9441ad16-3fce-11de-baaf-f74f6eef9491 [DEBUG] mod_nibblebill.c:395 do_billing() 0 seconds passed since last bill time of 2009-05-13 17:58:55 [DEBUG] mod_nibblebill.c:402 do_billing() Billing $0.036643 to 3 (Call: 9441ad16-3fce-11de-baaf-f74f6eef9491 / 0.000000 so far) On Wednesday 13 May 2009 16:46:37 Rupa Schomaker wrote: > Put them on the dialstring: > > application="bridge" data="{nibblevar1=val1,nibblevar2=val2}sofia/blah...." > > On Wed, May 13, 2009 at 5:32 AM, Yuriy Ivzhenko wrote: > > Hi All, > > > > I want to use mod_nibblebill for billing outgoing calls. > > > > My problem is billing begins before other paty answers the call. > > > > The WiKi tells: "You need to set the billing variables on your outbound > > calling leg and NOT on your A-Leg." > > > > How i can set nibblebill variables when B-leg connects? > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/8947b006/attachment.html From yudha2008 at gmail.com Thu May 14 02:42:57 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 14 May 2009 15:12:57 +0530 Subject: [Freeswitch-users] JavaScript session conference Message-ID: Hi, In JavaScript session i have one question: Step1: I have written one small JavaScript program first dial the one mobile number and one extension example: 9841798874====>1001 Step2: In that same JavaScript itself i want to transfer both the mobile number and extension into conference room 3001 Call Flow JavaScript session == (Mobile Number (9841799874) ==calls ==Extenson(1001)==calls ==Conference Room(3001)) *Javascript:* session = new Session(); session.originate(session, "sofia/default/sip:9841799874 at 192.168.1.135:5066 "); session.execute("bridge", "sofia/internal/1001 at 192.168.1.77"); session.execute("transfer", 9841799874); How can i use this ??? Session.execute("conference", conf_nr + "@default+" + memberflags); I want to transfer both the Extension and mobile Number in conference room in JavaScript session. How can i add it can any one assist me to solve this problem. Thanks in advance. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/6c26714b/attachment.html From saeedahmad1981 at gmail.com Thu May 14 02:46:56 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 14 May 2009 11:46:56 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: References: <2059908291DA4436B2009BBB52584C46@saeedlaptop><4A0AB9CF.4030307@telemaque.fr><2F7A3C109BB644DAAD92816534F1948F@saeedlaptop> Message-ID: <7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> Dear Mike, During ?Make Current? I see no errors, at the end I get successful installation message; I also tried to scroll up to see any possible errors but I guess there was nothing. I can see this message for mod_xml_cdr: making install mod_xml_cdr make[5]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' make[6]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' installing mod_xml_cdr.so make[6]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' make[5]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' but nothing for mod_xml_curl Again I copied mod_xml_curl.so to /mod directory and ?load mod_xml_curl? but it says load mod_xml_curl 2009-05-14 11:48:12 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object file: No such file or directory** API CALL [load(mod_xml_curl)] output: -ERR [module load file routine returned an error] Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Wednesday, May 13, 2009 7:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 do you see any errors if you do a make ? Mike On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote: Hi Tristan, No its not commented out. And strange thing is that it was working until I did ?make current? today. Thanks for your response. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tristan Sent: Wednesday, May 13, 2009 2:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 Hi Saeed, Maybe you did not built it ? ( see in modules.conf if it's not commented out with a # ) Saeed Ahmed a ?crit : Hi, I just did ?make current? and FS failed to load xml_curl, I looked into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy it from src directory but somehow FS doesn?t load it. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/037fcff5/attachment-0001.html From mrene_lists at avgs.ca Thu May 14 02:48:24 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 14 May 2009 11:48:24 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> References: <2059908291DA4436B2009BBB52584C46@saeedlaptop><4A0AB9CF.4030307@telemaque.fr><2F7A3C109BB644DAAD92816534F1948F@saeedlaptop> <7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> Message-ID: <4E945D5E-5DD0-4BF2-BEFE-C7220DE4126F@avgs.ca> try make mod_xml_curl-install Math On 14-May-09, at 11:46 AM, Saeed Ahmed wrote: > Dear Mike, > > During ?Make Current? I see no errors, at the end I get successful > installation message; I also tried to scroll up to see any possible > errors but I guess there was nothing. > > I can see this message for mod_xml_cdr: > > making install mod_xml_cdr > make[5]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > make[6]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > installing mod_xml_cdr.so > make[6]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > make[5]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > > but nothing for mod_xml_curl > > > Again I copied mod_xml_curl.so to /mod directory and ?load > mod_xml_curl? but it says > > load mod_xml_curl > 2009-05-14 11:48:12 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_xml_curl.so > **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared > object file: No such file or directory** > API CALL [load(mod_xml_curl)] output: > -ERR [module load file routine returned an error] > > Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Jerris > Sent: Wednesday, May 13, 2009 7:12 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk > 13288 > > do you see any errors if you do a make ? > > Mike > > On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote: > > > Hi Tristan, > > No its not commented out. > > And strange thing is that it was working until I did ?make current? > today. > > Thanks for your response. > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Tristan > Sent: Wednesday, May 13, 2009 2:15 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk > 13288 > > Hi Saeed, > > Maybe you did not built it ? > > ( see in modules.conf if it's not commented out with a # ) > > Saeed Ahmed a ?crit : > Hi, > > I just did ?make current? and FS failed to load xml_curl, I looked > into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy > it from src directory but somehow FS doesn?t load it. What could be > the problem? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/8149107e/attachment.html From jason at jasonjgw.net Thu May 14 03:11:22 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 14 May 2009 20:11:22 +1000 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> References: <7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> Message-ID: <20090514101122.GA7713@jdc.jasonjgw.net> Saeed Ahmed wrote: > During ?Make Current? I see no errors, at the end I get successful > installation message; I also tried to scroll up to see any possible errors > but I guess there was nothing. Have you done a fresh checkout and tried to rebuild from the beginning? When you do this, you should capture all of the output in a file so you can see what the errors are, e.g. make > build-log 2>&1 From saeedahmad1981 at gmail.com Thu May 14 03:52:29 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 14 May 2009 12:52:29 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <4E945D5E-5DD0-4BF2-BEFE-C7220DE4126F@avgs.ca> References: <2059908291DA4436B2009BBB52584C46@saeedlaptop><4A0AB9CF.4030307@telemaque.fr><2F7A3C109BB644DAAD92816534F1948F@saeedlaptop><7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> <4E945D5E-5DD0-4BF2-BEFE-C7220DE4126F@avgs.ca> Message-ID: <7A22030517784B4A97312AF5599AFAFC@saeedlaptop> Thanks Math, this has fixed the problem, but isn?t it strange that ?make current? is not doing it automatically? Many Thanks _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, May 14, 2009 11:48 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 try make mod_xml_curl-install Math On 14-May-09, at 11:46 AM, Saeed Ahmed wrote: Dear Mike, During ?Make Current? I see no errors, at the end I get successful installation message; I also tried to scroll up to see any possible errors but I guess there was nothing. I can see this message for mod_xml_cdr: making install mod_xml_cdr make[5]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' make[6]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' installing mod_xml_cdr.so make[6]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' make[5]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' but nothing for mod_xml_curl Again I copied mod_xml_curl.so to /mod directory and ?load mod_xml_curl? but it says load mod_xml_curl 2009-05-14 11:48:12 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object file: No such file or directory** API CALL [load(mod_xml_curl)] output: -ERR [module load file routine returned an error] Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Wednesday, May 13, 2009 7:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 do you see any errors if you do a make ? Mike On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote: Hi Tristan, No its not commented out. And strange thing is that it was working until I did ?make current? today. Thanks for your response. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tristan Sent: Wednesday, May 13, 2009 2:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 Hi Saeed, Maybe you did not built it ? ( see in modules.conf if it's not commented out with a # ) Saeed Ahmed a ?crit : Hi, I just did ?make current? and FS failed to load xml_curl, I looked into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy it from src directory but somehow FS doesn?t load it. What could be the problem? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/af5e418a/attachment-0001.html From mrene_lists at avgs.ca Thu May 14 03:56:19 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 14 May 2009 12:56:19 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: <7A22030517784B4A97312AF5599AFAFC@saeedlaptop> References: <2059908291DA4436B2009BBB52584C46@saeedlaptop><4A0AB9CF.4030307@telemaque.fr><2F7A3C109BB644DAAD92816534F1948F@saeedlaptop><7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop> <4E945D5E-5DD0-4BF2-BEFE-C7220DE4126F@avgs.ca> <7A22030517784B4A97312AF5599AFAFC@saeedlaptop> Message-ID: make current looks in modules.conf, check for typos.. and make sure there is no # in front of the line. You should see xml_int/mod_xml_curl Math On 14-May-09, at 12:52 PM, Saeed Ahmed wrote: > Thanks Math, this has fixed the problem, but isn?t it strange that > ?make current? is not doing it automatically? > > Many Thanks > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Mathieu Rene > Sent: Thursday, May 14, 2009 11:48 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk > 13288 > > try make mod_xml_curl-install > > Math > > On 14-May-09, at 11:46 AM, Saeed Ahmed wrote: > > > Dear Mike, > > During ?Make Current? I see no errors, at the end I get successful > installation message; I also tried to scroll up to see any possible > errors but I guess there was nothing. > > I can see this message for mod_xml_cdr: > > making install mod_xml_cdr > make[5]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > make[6]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > installing mod_xml_cdr.so > make[6]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > make[5]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/ > mod_xml_cdr' > > but nothing for mod_xml_curl > > > Again I copied mod_xml_curl.so to /mod directory and ?load > mod_xml_curl? but it says > > load mod_xml_curl > 2009-05-14 11:48:12 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_xml_curl.so > **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared > object file: No such file or directory** > API CALL [load(mod_xml_curl)] output: > -ERR [module load file routine returned an error] > > Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Jerris > Sent: Wednesday, May 13, 2009 7:12 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk > 13288 > > do you see any errors if you do a make ? > > Mike > > On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote: > > > > Hi Tristan, > > No its not commented out. > > And strange thing is that it was working until I did ?make current? > today. > > Thanks for your response. > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Tristan > Sent: Wednesday, May 13, 2009 2:15 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk > 13288 > > Hi Saeed, > > Maybe you did not built it ? > > ( see in modules.conf if it's not commented out with a # ) > > Saeed Ahmed a ?crit : > Hi, > > I just did ?make current? and FS failed to load xml_curl, I looked > into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy > it from src directory but somehow FS doesn?t load it. What could be > the problem? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/e4396827/attachment-0001.html From saeedahmad1981 at gmail.com Thu May 14 04:14:27 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 14 May 2009 13:14:27 +0200 Subject: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 In-Reply-To: References: <2059908291DA4436B2009BBB52584C46@saeedlaptop><4A0AB9CF.4030307@telemaque.fr><2F7A3C109BB644DAAD92816534F1948F@saeedlaptop><7F863A5ED45F4DD484ED6EC1F19D5868@saeedlaptop><4E945D5E-5DD0-4BF2-BEFE-C7220DE4126F@avgs.ca><7A22030517784B4A97312AF5599AFAFC@saeedlaptop> Message-ID: Yes it was commented in src directory, I don?t know why? Because before it was working OK and I never changed it by myself. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, May 14, 2009 12:56 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 make current looks in modules.conf, check for typos.. and make sure there is no # in front of the line. You should see xml_int/mod_xml_curl Math On 14-May-09, at 12:52 PM, Saeed Ahmed wrote: Thanks Math, this has fixed the problem, but isn?t it strange that ?make current? is not doing it automatically? Many Thanks _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, May 14, 2009 11:48 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 try make mod_xml_curl-install Math On 14-May-09, at 11:46 AM, Saeed Ahmed wrote: Dear Mike, During ?Make Current? I see no errors, at the end I get successful installation message; I also tried to scroll up to see any possible errors but I guess there was nothing. I can see this message for mod_xml_cdr: making install mod_xml_cdr make[5]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' make[6]: Entering directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' installing mod_xml_cdr.so make[6]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' make[5]: Leaving directory `/usr/src/freeswitch-svn/src/mod/xml_int/mod_xml_cdr' but nothing for mod_xml_curl Again I copied mod_xml_curl.so to /mod directory and ?load mod_xml_curl? but it says load mod_xml_curl 2009-05-14 11:48:12 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **/usr/local/freeswitch/mod/mod_xml_curl.so: cannot open shared object file: No such file or directory** API CALL [load(mod_xml_curl)] output: -ERR [module load file routine returned an error] Thanks. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Wednesday, May 13, 2009 7:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 do you see any errors if you do a make ? Mike On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote: Hi Tristan, No its not commented out. And strange thing is that it was working until I did ?make current? today. Thanks for your response. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tristan Sent: Wednesday, May 13, 2009 2:15 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod_xml_curl doesn't exists Trunk 13288 Hi Saeed, Maybe you did not built it ? ( see in modules.conf if it's not commented out with a # ) Saeed Ahmed a ?crit : Hi, I just did ?make current? and FS failed to load xml_curl, I looked into mod/ dir but mod_xml_curl.so wasn?t there. I even tried to copy it from src directory but somehow FS doesn?t load it. What could be the problem? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/c03acdec/attachment-0001.html From kjv at ken-ton.com Thu May 14 06:06:22 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Thu, 14 May 2009 09:06:22 -0400 Subject: [Freeswitch-users] JavaScript session conference In-Reply-To: References: Message-ID: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> While this doesn't explain how to do it in Javascript, it will probably give you an idea: http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On May 14, 2009, at 5:42 AM, Baskar wrote: > Hi, > > In JavaScript session i have one question: > > Step1: I have written one small JavaScript program first dial the > one mobile number and one extension example: 9841798874====>1001 > > Step2: In that same JavaScript itself i want to transfer both the > mobile number and extension into conference room 3001 > > Call Flow > > JavaScript session == (Mobile Number (9841799874) ==calls > ==Extenson(1001)==calls ==Conference Room(3001)) > > Javascript: > > session = new Session(); > session.originate(session, "sofia/default/sip:9841799874 at 192.168.1.135:5066 > "); > session.execute("bridge", "sofia/internal/1001 at 192.168.1.77"); > session.execute("transfer", 9841799874); > > How can i use this ??? > > Session.execute("conference", conf_nr + "@default+" + memberflags); > > I want to transfer both the Extension and mobile Number in > conference room in JavaScript session. How can i add it can any one > assist me to solve this problem. > > > > Thanks in advance. > -- > Warm Regards, > N.Baskar > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/2dfc1ce5/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/2dfc1ce5/attachment.bin From yudha2008 at gmail.com Thu May 14 06:55:41 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 14 May 2009 19:25:41 +0530 Subject: [Freeswitch-users] JavaScript session conference In-Reply-To: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> References: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> Message-ID: Hi, Karl Vesterling Thanks for the reply But i need the above process to be done through javascript session. Can some one assist me to solve this problem. Thanks in advance. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/d1cdc6e2/attachment.html From Mailings at kh-dev.de Wed May 13 15:55:05 2009 From: Mailings at kh-dev.de (Mailings) Date: Thu, 14 May 2009 00:55:05 +0200 Subject: [Freeswitch-users] t38modem + FreeSWITCH problem Message-ID: Hi, I just tried FreeSWITCH together with t38modem and HylaFAX. But when I bridge the call to the modem I get the following error: 2009-05-14 00:22:38 [ERR] sofia.c:3280 sofia_handle_sip_i_state() Reinvite Codec Error! Does anyone know how to set this up? To PSTN I use a Lancom router which converts T38 to Euro-ISDN and vice versa. The t38modem was started with the following parameters: /usr/bin/t38modem -tt -o /var/log/t38modem.log --no-h323 -u t38modem --sip-register t38modem at freeswitch,password --sip-listen udp\$127.0.0.1:6060 --ptty +/dev/ttyT38-1 --route "modem:.*=sip:@127.0.0.1" --route "sip:.*=modem:" I also created a directory entry for FreeSWITCH with the following content: The bridging is done in perl: $session->execute("bridge","user/t38modem@" . $VARS{"domain"}); Any help is appreciated. Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/7dca3044/attachment-0001.html From vijay11tiwari at hotmail.com Thu May 14 07:26:40 2009 From: vijay11tiwari at hotmail.com (vijay tiwari) Date: Thu, 14 May 2009 19:56:40 +0530 Subject: [Freeswitch-users] context is not working Message-ID: hello all i have created a two user one with defualt user_context and other one with which i created with the name "US". and i am able to make a call when using defualt user_context. but when i try to call from user which have user_context="US" the is not working, logs when call is working 2009-05-13 20:54:45 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/US/8989 at 122.160.83.216 SOFIA ROUTING 2009-05-13 20:54:45 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/US/8989 at 122.160.83.216 Standard ROUTING 2009-05-13 20:54:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 8989->01112127773456 in context default 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [unloop] ${unroll_loops}(true) =~ /^true$/ 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [unloop] ${sip_looped_call}() =~ /^true$/ . . . 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [XXX] destination_number(01112127773456) =~ /^011(\d+)$/ 2009-05-13 20:54:45 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/US/8989 at 122.160.83.216) State Change CS_ROUTING -> CS_EXECUTE logs when call not working 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (sofia/US/12345 at 122.160.83.216) State ROUTING 2009-05-13 20:22:27 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/US/12345 at 122.160.83.216 SOFIA ROUTING 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/US/12345 at 122.160.83.216 Standard ROUTING 2009-05-13 20:22:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 12345->01112127773456 in context US 2009-05-13 20:22:27 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-05-13 20:22:27 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/US/12345 at 122.160.83.216 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-05-13 20:22:27 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/US/12345 at 122.160.83.216 [KILL] 2009-05-13 20:22:27 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/US/12345 at 122.160.83.216 [BREAK] 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (sofia/US/12345 at 122.160.83.216) State ROUTING going to sleep 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/US/12345 at 122.160.83.216) Running State Change CS_HANGUP _________________________________________________________________ Live Search extreme As India feels the heat of poll season, get all the info you need on the MSN News Aggregator http://news.in.msn.com/National/indiaelections2009/aggregator/default.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/3eb93e29/attachment-0001.html From mrene_lists at avgs.ca Thu May 14 07:30:09 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 14 May 2009 16:30:09 +0200 Subject: [Freeswitch-users] context is not working In-Reply-To: References: Message-ID: <7E62A325-D0BA-40CF-AF9D-66EA37A67813@avgs.ca> Please paste the user directory entry. Math On 14-May-09, at 4:26 PM, vijay tiwari wrote: > hello all > > i have created a two user one with defualt user_context and other > one with which i created with the name "US". and i am able to make > a call when using defualt user_context. but when i try to call from > user which have user_context="US" the is not working, > > logs when call is working > > 2009-05-13 20:54:45 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/US/8989 at 122.160.83.216 > SOFIA ROUTING > 2009-05-13 20:54:45 [DEBUG] switch_core_state_machine.c:64 > switch_core_standard_on_routing() sofia/US/8989 at 122.160.83.216 > Standard ROUTING > 2009-05-13 20:54:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 8989->01112127773456 in context default > 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() > Regex: [unloop] ${unroll_loops}(true) =~ /^true$/ > 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() > Regex: [unloop] ${sip_looped_call}() =~ /^true$/ > . > . > . > 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() > Regex: [XXX] destination_number(01112127773456) =~ /^011(\d+)$/ > 2009-05-13 20:54:45 [DEBUG] switch_core_state_machine.c:100 > switch_core_standard_on_routing() (sofia/US/8989 at 122.160.83.216) > State Change CS_ROUTING -> CS_EXECUTE > > > > logs when call not working > > 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (sofia/US/12345 at 122.160.83.216) State > ROUTING > 2009-05-13 20:22:27 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/US/12345 at 122.160.83.216 > SOFIA ROUTING > 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:64 > switch_core_standard_on_routing() sofia/US/12345 at 122.160.83.216 > Standard ROUTING > 2009-05-13 20:22:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 12345->01112127773456 in context US > 2009-05-13 20:22:27 [INFO] switch_core_state_machine.c:122 > switch_core_standard_on_routing() No Route, Aborting > 2009-05-13 20:22:27 [NOTICE] switch_core_state_machine.c:123 > switch_core_standard_on_routing() Hangup sofia/US/ > 12345 at 122.160.83.216 [CS_ROUTING] [NO_ROUTE_DESTINATION] > 2009-05-13 20:22:27 [DEBUG] switch_channel.c:1566 > switch_channel_perform_hangup() Send signal sofia/US/12345 at 122.160.83.216 > [KILL] > 2009-05-13 20:22:27 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/US/12345 at 122.160.83.216 > [BREAK] > 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (sofia/US/12345 at 122.160.83.216) State > ROUTING going to sleep > 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/US/12345 at 122.160.83.216) Running > State Change CS_HANGUP > > > Get easy photo sharing with Windows LiveT Photos. Drag n' drop > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/8ad68360/attachment.html From intralanman at freeswitch.org Thu May 14 07:37:51 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 14 May 2009 10:37:51 -0400 Subject: [Freeswitch-users] context is not working In-Reply-To: <7E62A325-D0BA-40CF-AF9D-66EA37A67813@avgs.ca> References: <7E62A325-D0BA-40CF-AF9D-66EA37A67813@avgs.ca> Message-ID: <4A0C2CBF.1010101@freeswitch.org> The dialplan for the context US might be helpful also -Ray Mathieu Rene wrote: > Please paste the user directory entry. > > Math > > On 14-May-09, at 4:26 PM, vijay tiwari wrote: > >> hello all >> >> i have created a two user one with defualt user_context and other one >> with which i created with the name "US". and i am able to make a >> call when using defualt user_context. but when i try to call from >> user which have user_context="US" the is not working, >> >> logs when call is working >> >> 2009-05-13 20:54:45 [DEBUG] mod_sofia.c:130 >> sofia_on_routing() sofia/US/8989 at 122.160.83.216 >> SOFIA ROUTING >> 2009-05-13 20:54:45 [DEBUG] switch_core_state_machine.c:64 >> switch_core_standard_on_routing() sofia/US/8989 at 122.160.83.216 >> Standard ROUTING >> 2009-05-13 20:54:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 8989->01112127773456 in context default >> 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: [unloop] ${unroll_loops}(true) =~ /^true$/ >> 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: [unloop] ${sip_looped_call}() =~ /^true$/ >> . >> . >> . >> 2009-05-13 20:54:45 [DEBUG] mod_dialplan_xml.c:117 parse_exten() >> Regex: [XXX] destination_number(01112127773456) =~ /^011(\d+)$/ >> 2009-05-13 20:54:45 [DEBUG] switch_core_state_machine.c:100 >> switch_core_standard_on_routing() (sofia/US/8989 at 122.160.83.216 >> ) State Change CS_ROUTING -> >> CS_EXECUTE >> >> >> >> logs when call not working >> >> 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:457 >> switch_core_session_run() (sofia/US/12345 at 122.160.83.216 >> ) State ROUTING >> 2009-05-13 20:22:27 [DEBUG] mod_sofia.c:130 >> sofia_on_routing() sofia/US/12345 at 122.160.83.216 >> SOFIA ROUTING >> 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:64 >> switch_core_standard_on_routing() sofia/US/12345 at 122.160.83.216 >> Standard ROUTING >> 2009-05-13 20:22:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing 12345->01112127773456 in context US >> 2009-05-13 20:22:27 [INFO] switch_core_state_machine.c:122 >> switch_core_standard_on_routing() No Route, Aborting >> 2009-05-13 20:22:27 [NOTICE] switch_core_state_machine.c:123 >> switch_core_standard_on_routing() >> Hangup sofia/US/12345 at 122.160.83.216 >> [CS_ROUTING] >> [NO_ROUTE_DESTINATION] >> 2009-05-13 20:22:27 [DEBUG] switch_channel.c:1566 >> switch_channel_perform_hangup() Send >> signal sofia/US/12345 at 122.160.83.216 >> [KILL] >> 2009-05-13 20:22:27 [DEBUG] switch_core_session.c:820 >> switch_core_session_signal_state_change() Send >> signal sofia/US/12345 at 122.160.83.216 >> [BREAK] >> 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:457 >> switch_core_session_run() (sofia/US/12345 at 122.160.83.216 >> ) State ROUTING going to sleep >> 2009-05-13 20:22:27 [DEBUG] switch_core_state_machine.c:383 >> switch_core_session_run() (sofia/US/12345 at 122.160.83.216 >> ) Running State Change CS_HANGUP >> >> >> ------------------------------------------------------------------------ >> Get easy photo sharing with Windows LiveT Photos. Drag n' drop >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/75241617/attachment.html From mfedyk at mikefedyk.com Thu May 14 09:32:56 2009 From: mfedyk at mikefedyk.com (Mike Fedyk) Date: Thu, 14 May 2009 09:32:56 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> Message-ID: <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> Why is initiating a conference from a dial plan entry better than directly from an internal script? Is there a page that details what is and isn't script abuse? On 5/5/09, Anthony Minessale wrote: > You should rule out the network problems first, which sound more likely. > > you can reduce the overuse of JS if you transfer the call to a regular > extension with a dynamic regex. > > session.execute("transfer", "conf-xyz"); > > then make a regex in your xml dialplan to pick up ^conf-(.*) and execute > conference $1 > > > On Tue, May 5, 2009 at 9:48 AM, Stephen Crosby wrote: > >> Forgot to add that my OS is Ubuntu 8.04LTS (hardy heron). >> >> --Stephen >> >> >> On Tue, May 5, 2009 at 7:42 AM, Stephen Crosby >> wrote: >> >>> I know I'm not on svn trunk, but this is a production server and it's >>> just >>> not feasible to update it constantly. I can update it though if you think >>> I >>> need to. I am routing callers to the conference app with javascript like >>> this: >>> session.execute("conference", xyz); >>> Can you tell me more about the problems I could have? >>> >>> The machine running freeswitch has 1024MB memory and I'm not sure about >>> the CPU since its a VPS. >>> >>> --Stephen >>> >>> On Tue, May 5, 2009 at 3:49 AM, Brian West wrote: >>> >>>> First off you're not on SVN trunk secondly Are you executing the >>>> conference app inside your js file? If so then there could be the >>>> problem! >>>> You have also forgotten to include anything about Distro, OS, CPU and >>>> Memory. >>>> /b >>>> >>>> On May 5, 2009, at 12:12 AM, Stephen Crosby wrote: >>>> >>>> We had our first big issues with our freeswitch system today. During at >>>> least 2 conferences, audio became jittery and there were three occasions >>>> where everyone was dropped from a conference. Even so, conference >>>> recording >>>> was not interrupted, and the freeswitch debug log doesn't show anything >>>> unusual. >>>> >>>> Our hardware monitoring software doesn't show any kind of unusual >>>> resources usage, and our web host claims there were no outages during >>>> the >>>> time when we were experiencing problems. >>>> >>>> We're currently running revision 12259. >>>> >>>> How should I proceed in diagnosing this issue? >>>> >>>> --Stephen >>>> >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Sent from my mobile device From msc at freeswitch.org Thu May 14 09:50:04 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 May 2009 09:50:04 -0700 Subject: [Freeswitch-users] t38modem + FreeSWITCH problem In-Reply-To: References: Message-ID: <87f2f3b90905140950k422ef374qd54f43df324b3e72@mail.gmail.com> I believe we talked about this in the channel last night... Did you try proxy-media mode? -MC On Wed, May 13, 2009 at 3:55 PM, Mailings wrote: > Hi, > > > > I just tried FreeSWITCH together with t38modem and HylaFAX. > > But when I bridge the call to the modem I get the following error: > > > > 2009-05-14 00:22:38 [ERR] sofia.c:3280 sofia_handle_sip_i_state() Reinvite > Codec Error! > > > > Does anyone know how to set this up? > > > > To PSTN I use a Lancom router which converts T38 to Euro-ISDN and vice > versa. > > > > The t38modem was started with the following parameters: > > /usr/bin/t38modem -tt -o /var/log/t38modem.log --no-h323 -u t38modem > --sip-register t38modem at freeswitch,password --sip-listen udp\$ > 127.0.0.1:6060 --ptty +/dev/ttyT38-1 --route "modem:.*=sip:@127.0.0.1" > --route "sip:.*=modem:" > > > > I also created a directory entry for FreeSWITCH with the following content: > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > > value="$${outbound_caller_id}"/> > > > > > > > > > > > > > > The bridging is done in perl: > > $session->execute("bridge","user/t38modem@" . $VARS{"domain"}); > > > > Any help is appreciated. > > > > Thanks, Klaus > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/68ca2156/attachment.html From msc at freeswitch.org Thu May 14 09:55:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 May 2009 09:55:31 -0700 Subject: [Freeswitch-users] JavaScript session conference In-Reply-To: References: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> Message-ID: <87f2f3b90905140955y6758fdb5w61470affd43496e9@mail.gmail.com> Can you do each operation sequentially? I mean, is it a requirement that the two parties be bridged together prior to going into the conference room? I'm curious about the sequence of the calls. Also, what is the big picture of this application? One caller calls from a mobile phone into extension 1001 and if 1001 answers then both parties get put into conference room 3001? -MC On Thu, May 14, 2009 at 6:55 AM, Baskar wrote: > Hi, > Karl Vesterling Thanks for the reply But i need the above process to be > done through javascript session. > Can some one assist me to solve this problem. > Thanks in advance. > -- > Warm Regards, > N.Baskar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/098d94d4/attachment.html From krice at freeswitch.org Thu May 14 09:56:11 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 14 May 2009 11:56:11 -0500 Subject: [Freeswitch-users] t38modem + FreeSWITCH problem In-Reply-To: <87f2f3b90905140950k422ef374qd54f43df324b3e72@mail.gmail.com> Message-ID: You need to be in proxy media mode... FreeSWITCH does not have a full udptl proxy at this time so its either bypass media modes or proxy media mode only From: Michael Collins Reply-To: Date: Thu, 14 May 2009 09:50:04 -0700 To: Subject: Re: [Freeswitch-users] t38modem + FreeSWITCH problem I believe we talked about this in the channel last night... Did you try proxy-media mode? -MC On Wed, May 13, 2009 at 3:55 PM, Mailings wrote: > Hi, > > ? > > I just tried FreeSWITCH together with t38modem and HylaFAX. > > But when I bridge the call to the modem I get the following error: > > ? > > 2009-05-14 00:22:38 [ERR] sofia.c:3280 sofia_handle_sip_i_state() Reinvite > Codec Error! > > ? > > Does anyone know how to set this up? > > ? > > To PSTN I use a Lancom router which converts T38 to Euro-ISDN and vice versa. > > ? > > The t38modem was started with the following parameters: > > ? /usr/bin/t38modem -tt -o /var/log/t38modem.log --no-h323 -u t38modem > --sip-register t38modem at freeswitch,password --sip-listen udp\$127.0.0.1:6060 > --ptty +/dev/ttyT38-1 --route > "modem:.*=sip:@127.0.0.1 " --route "sip:.*=modem:" > > ? > > I also created a directory entry for FreeSWITCH with the following content: > > ? > > ? ?? > > ??? ?? > > ????? ?? > > ? ???? > > ? ???? > > ??? ???? > > ????? ?? > > ? ?????? > > ? ?????? > > ??? ???? > > ? ?????? value="$${outbound_caller_name}"/> > > ? ?????? value="$${outbound_caller_id}"/> > > ??? ???? > > ????? ?? > > ? ???? > > ? ?? > > ? > > ? > > The bridging is done in perl: > > ? $session->execute("bridge","user/t38modem@" . $VARS{"domain"}); > > ? > > Any help is appreciated. > > ? > > Thanks, Klaus > > ? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/c4a59683/attachment.html From msc at freeswitch.org Thu May 14 09:58:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 14 May 2009 09:58:30 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> Message-ID: <87f2f3b90905140958i18215182vaed426193c43bd04@mail.gmail.com> Javascript has it's own issues. Tony wrote FreeSWITCH, including the dialplan parser. The less you can use JS, especially in a high-volume environment, the better. The happy medium is to use Lua. It is small, lightweight, easy, and designed to be embedded. Tony has had 3000 Lua session up simultaneously with no issues. I guarantee you that JS would have a meltdown at a fraction of that volume... -MC On Thu, May 14, 2009 at 9:32 AM, Mike Fedyk wrote: > Why is initiating a conference from a dial plan entry better than > directly from an internal script? Is there a page that details what > is and isn't script abuse? > > On 5/5/09, Anthony Minessale wrote: > > You should rule out the network problems first, which sound more likely. > > > > you can reduce the overuse of JS if you transfer the call to a regular > > extension with a dynamic regex. > > > > session.execute("transfer", "conf-xyz"); > > > > then make a regex in your xml dialplan to pick up ^conf-(.*) and execute > > conference $1 > > > > > > On Tue, May 5, 2009 at 9:48 AM, Stephen Crosby > wrote: > > > >> Forgot to add that my OS is Ubuntu 8.04LTS (hardy heron). > >> > >> --Stephen > >> > >> > >> On Tue, May 5, 2009 at 7:42 AM, Stephen Crosby > >> wrote: > >> > >>> I know I'm not on svn trunk, but this is a production server and it's > >>> just > >>> not feasible to update it constantly. I can update it though if you > think > >>> I > >>> need to. I am routing callers to the conference app with javascript > like > >>> this: > >>> session.execute("conference", xyz); > >>> Can you tell me more about the problems I could have? > >>> > >>> The machine running freeswitch has 1024MB memory and I'm not sure about > >>> the CPU since its a VPS. > >>> > >>> --Stephen > >>> > >>> On Tue, May 5, 2009 at 3:49 AM, Brian West > wrote: > >>> > >>>> First off you're not on SVN trunk secondly Are you executing the > >>>> conference app inside your js file? If so then there could be the > >>>> problem! > >>>> You have also forgotten to include anything about Distro, OS, CPU and > >>>> Memory. > >>>> /b > >>>> > >>>> On May 5, 2009, at 12:12 AM, Stephen Crosby wrote: > >>>> > >>>> We had our first big issues with our freeswitch system today. During > at > >>>> least 2 conferences, audio became jittery and there were three > occasions > >>>> where everyone was dropped from a conference. Even so, conference > >>>> recording > >>>> was not interrupted, and the freeswitch debug log doesn't show > anything > >>>> unusual. > >>>> > >>>> Our hardware monitoring software doesn't show any kind of unusual > >>>> resources usage, and our web host claims there were no outages during > >>>> the > >>>> time when we were experiencing problems. > >>>> > >>>> We're currently running revision 12259. > >>>> > >>>> How should I proceed in diagnosing this issue? > >>>> > >>>> --Stephen > >>>> > >>>> > >>>> Brian West > >>>> brian at freeswitch.org > >>>> > >>>> -- Meet us at ClueCon! http://www.cluecon.com > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > -- > Sent from my mobile device > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/8f68549e/attachment-0001.html From brian at freeswitch.org Thu May 14 10:01:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 14 May 2009 12:01:20 -0500 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> Message-ID: The script has a javascript interpreter allocated sitting there eating up ram for no reason... so to scale better its wise to exit the script and do your conference. Lua is lighter weight but still having the interpreter hang around when not needed isn't wise. /b On May 14, 2009, at 11:32 AM, Mike Fedyk wrote: > Why is initiating a conference from a dial plan entry better than > directly from an internal script? Is there a page that details what > is and isn't script abuse? > > On 5/5/09, Anthony Minessale wrote: >> You should rule out the network problems first, which sound more >> likely. >> >> you can reduce the overuse of JS if you transfer the call to a >> regular >> extension with a dynamic regex. >> >> session.execute("transfer", "conf-xyz"); >> >> then make a regex in your xml dialplan to pick up ^conf-(.*) and >> execute >> conference $1 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/02b293b0/attachment.html From stevecrozz at gmail.com Thu May 14 10:24:35 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 14 May 2009 10:24:35 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> Message-ID: <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> I keep hearing that lua is lighter weight / more scalable than javascript. I'd love to see some data that shows how big the difference really is. I could port all my scripts from javascript to lua, but it would require a lot of overhead (me learning lua + actually porting the scripts). I want to know how much benefit there really is to that. --Stephen On Thu, May 14, 2009 at 10:01 AM, Brian West wrote: > The script has a javascript interpreter allocated sitting there eating up > ram for no reason... so to scale better its wise to exit the script and do > your conference. > Lua is lighter weight but still having the interpreter hang around when not > needed isn't wise. > > /b > > On May 14, 2009, at 11:32 AM, Mike Fedyk wrote: > > Why is initiating a conference from a dial plan entry better than > directly from an internal script? Is there a page that details what > is and isn't script abuse? > > On 5/5/09, Anthony Minessale wrote: > > You should rule out the network problems first, which sound more likely. > > > you can reduce the overuse of JS if you transfer the call to a regular > > extension with a dynamic regex. > > > session.execute("transfer", "conf-xyz"); > > > then make a regex in your xml dialplan to pick up ^conf-(.*) and execute > > conference $1 > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/56c4cec4/attachment.html From brian at freeswitch.org Thu May 14 10:31:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 14 May 2009 12:31:05 -0500 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> Message-ID: Its obvious if you look at the size of the JS VM vs the lua VM.. it would clearly scale better not needing megs and megs of ram per call vs about 160kb for lua. Lua took about 20 min to learn and about an hour to get the finer points down. /b On May 14, 2009, at 12:24 PM, Stephen Crosby wrote: > I keep hearing that lua is lighter weight / more scalable than > javascript. I'd love to see some data that shows how big the > difference really is. I could port all my scripts from javascript to > lua, but it would require a lot of overhead (me learning lua + > actually porting the scripts). I want to know how much benefit there > really is to that. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/176be9a5/attachment.html From niall.crosby at gmail.com Thu May 14 10:37:43 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Thu, 14 May 2009 18:37:43 +0100 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> Message-ID: <4aec92830905141037s50550e1cq6bb987b077fdc1ad@mail.gmail.com> How would using the mod_event_socket compare to lua / javascript with regards performance and scaling? (ignoring the client side of the socket, just the Freeswitch side). Is there any performance issues to placing a call into conference using with callflow from the mod_socket? N. 2009/5/14 Brian West : > Its obvious if you look at the size of the JS VM vs the lua VM.. it would > clearly scale better not needing megs and megs of ram per call vs about > 160kb for lua. ?Lua took about 20 min to learn and about an hour to get the > finer points down. > /b > > On May 14, 2009, at 12:24 PM, Stephen Crosby wrote: > > I keep hearing that lua is lighter weight / more scalable than javascript. > I'd love to see some data that shows how big the difference really is. I > could port all my scripts from javascript to lua, but it would require a lot > of overhead (me learning lua + actually porting the scripts). I want to know > how much benefit there really is to that. > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- Sremium Ltd. Reg Number: 451937 Mobile: +353 (0)87 2393174 Web: www.sremium.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of Sremium. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. From marketing at cluecon.com Thu May 14 13:14:41 2009 From: marketing at cluecon.com (Michael Collins) Date: Thu, 14 May 2009 13:14:41 -0700 Subject: [Freeswitch-users] Call For Participants: Lightning Talks at ClueCon 2009 Message-ID: <87f2f3b90905141314w3a2b24ccu41c692588555264b@mail.gmail.com> *ClueCon 2009 is coming soon!* We are interested in your thoughts on subjects for lighting talks. We would love to have a number of 5-10 minute presentations by members of the community. If you would like to give a talk, or just have an idea for a talk, please let us know. How do lightning talks work? Quite simply, the presenter has just a few minutes to speak on a particular subject, usually no more than 10 minutes. He or she will deliver the information rapidly, which means keeping the presentation focused tightly on the subject being discussed. Lightning talks usually do not have enough time for audience Q&A. However, ClueCon has a long lunch period that is designed to allow attendees plenty of time to interact. Those are perfect times to discuss lightning talks or any other presentations. Those who give presentations enjoy interacting with other attendees in a relaxed atmosphere during lunch or in the evening at dinner. If you haven't already registered for ClueCon 2009 then please call us at 877.742.CLUE right away and we will complete your registration. Also, don't forget that expedia.com has some nice hotel deals for the Wyndham Chicago. Book your room today! We look forward to hearing from you and seeing you all at ClueCon in Chicago. -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/e7c093e9/attachment-0001.html From larclap at yahoo.com Thu May 14 13:21:55 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 14 May 2009 13:21:55 -0700 Subject: [Freeswitch-users] Help configuring external profile In-Reply-To: <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> Message-ID: <001101c9d4d1$a0914c60$e1b3e520$@com> I am a total newbie. I have set up FS on CentOS with this group's help. I can make an outbound call via an ITSP (flowroute). I have struggled receiving a call. Freeswitch does not answer when I dial the DID from outside. I already have an Asterisk box on the LAN with SIP port at 5060. So I asked the group what to do and the reply was to change the sip port in vars.xml, which I did. But ignorantly (I think) I set the internal_sip_port=5090, rather than the external_sip_port. It dawned on me that I might have made this mistake when I look at: freeswitch at fs> sofia status API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia at 192.168.10.29:5090 RUNNING (0) external profile sip:mod_sofia at 64.105.128.82:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG flowroute gateway sip:77505382 at sip.flowroute.com REGED 192.168.10.29 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5090 RUNNING (0) default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED ============================================================================ ===================== 3 profiles 4 aliases So if I change external_sip_port=5090, I think I should be OK. However, the 64.105.128.82 is not correct. Since I had to create a new SIP port at 5090, I had to tell my hardware firewall about it. So I created SIP=5090 at 64.105.128.83 (the Asterisk box is sitting at 64.105.128.82). How is FS getting 64.105.128.82 in 'sofia status' and how can I change that to 64.105.128.83? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/87fc2344/attachment.html From brian at freeswitch.org Thu May 14 13:27:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 14 May 2009 15:27:06 -0500 Subject: [Freeswitch-users] Help configuring external profile In-Reply-To: <001101c9d4d1$a0914c60$e1b3e520$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> <001101c9d4d1$a0914c60$e1b3e520$@com> Message-ID: <13B3E523-EC28-4E48-86D9-CBB30412FFA8@freeswitch.org> Its guessing the IP you'll have to set it to the IP you want by opening up the sofia profile in sip_profiles/internal.xml and external.xml and specify the EXACT IP to use. On another note please stop hijacking threads. /b On May 14, 2009, at 3:21 PM, Lars Zeb wrote: > How is FS getting 64.105.128.82 in ?sofia status? and how can I > change that to 64.105.128.83? > > Thanks, Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/035a69e7/attachment.html From stevecrozz at gmail.com Thu May 14 13:36:17 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 14 May 2009 13:36:17 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> Message-ID: <11990ade0905141336r1e499d4i15e2df87ecd2e6cb@mail.gmail.com> I think its less obvious to a user who'd never heard of lua until recently. I think it would be very helpful if someone had a comparison and could put it on the wiki. I spent some time poking around on the wiki before I decided to write my scripts in javascript. If I had seen any warnings on scalability/performance, especially with good data to back it up, I probably would have chosen something else. --Stephen On Thu, May 14, 2009 at 10:31 AM, Brian West wrote: > Its obvious if you look at the size of the JS VM vs the lua VM.. it would > clearly scale better not needing megs and megs of ram per call vs about > 160kb for lua. Lua took about 20 min to learn and about an hour to get the > finer points down. > /b > > > On May 14, 2009, at 12:24 PM, Stephen Crosby wrote: > > I keep hearing that lua is lighter weight / more scalable than javascript. > I'd love to see some data that shows how big the difference really is. I > could port all my scripts from javascript to lua, but it would require a lot > of overhead (me learning lua + actually porting the scripts). I want to know > how much benefit there really is to that. > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/3d319679/attachment.html From larclap at yahoo.com Thu May 14 13:39:10 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 14 May 2009 13:39:10 -0700 Subject: [Freeswitch-users] Help configuring external profile In-Reply-To: <13B3E523-EC28-4E48-86D9-CBB30412FFA8@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> <001101c9d4d1$a0914c60$e1b3e520$@com> <13B3E523-EC28-4E48-86D9-CBB30412FFA8@freeswitch.org> Message-ID: <004301c9d4d4$08f1f640$1ad5e2c0$@com> Sorry about hijacking, but please tell me how I did that so I can avoid it in the future. I merely sent a message to freeswitch-users at lists.freeswitch.org with a subject of 'Help configuring external profile'. If there's another to post a question, I'd be glad to do it. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, May 14, 2009 1:27 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help configuring external profile Its guessing the IP you'll have to set it to the IP you want by opening up the sofia profile in sip_profiles/internal.xml and external.xml and specify the EXACT IP to use. On another note please stop hijacking threads. /b On May 14, 2009, at 3:21 PM, Lars Zeb wrote: How is FS getting 64.105.128.82 in 'sofia status' and how can I change that to 64.105.128.83? Thanks, Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/155dc622/attachment-0001.html From brian at freeswitch.org Thu May 14 13:43:13 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 14 May 2009 15:43:13 -0500 Subject: [Freeswitch-users] Help configuring external profile In-Reply-To: <004301c9d4d4$08f1f640$1ad5e2c0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <383217D3-C6FA-413B-8738-74453A487E65@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363835@cooper> <63C8503A-52F7-4D9E-AA30-7C0FA1480A00@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB636383E@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> <001101c9d4d1$a0914c60$e1b3e520$@com> <13B3E523-EC28-4E48-86D9-CBB30412FFA8@freeswitch.org> <004301c9d4d4$08f1f640$1ad5e2c0$@com> Message-ID: <9DF9CFF1-D919-4168-B3B5-1B2A6E551274@freeswitch.org> Make sure you click new message and input the address... If you click reply.. change the subject and the body thats how you end up hijacking threads! ;) /b On May 14, 2009, at 3:39 PM, Lars Zeb wrote: > Sorry about hijacking, but please tell me how I did that so I can > avoid it in the future. I merely sent a message to freeswitch-users at lists.freeswitch.org > with a subject of ?Help configuring external profile?. If there?s > another to post a question, I?d be glad to do it. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/57ebd97b/attachment.html From pete at privateconnect.com Thu May 14 14:22:37 2009 From: pete at privateconnect.com (pete at privateconnect.com) Date: Thu, 14 May 2009 14:22:37 -0700 Subject: [Freeswitch-users] =?utf-8?q?help_with_mod=5Fconference_stability?= Message-ID: <20090514142237.2ad02225396a31c9de30536f2e338977.d94bc79c17.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/a80efee3/attachment.html From stevecrozz at gmail.com Thu May 14 14:34:57 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 14 May 2009 14:34:57 -0700 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <20090514142237.2ad02225396a31c9de30536f2e338977.d94bc79c17.wbe@email04.secureserver.net> References: <20090514142237.2ad02225396a31c9de30536f2e338977.d94bc79c17.wbe@email04.secureserver.net> Message-ID: <11990ade0905141434o643670e8r45adf9aa47e438db@mail.gmail.com> Thanks Pete, I've been hanging out on #freeswitch and I started a conversation on this topic. User 'mercutioviz' created this wiki page mercutioviz: http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F which should help new freeswitchers not make the mistake we made and have to rewrite everything. If you have anything to add to this, it would be appreciated I'm sure. --Stephen On Thu, May 14, 2009 at 2:22 PM, wrote: > > I will further confirm Brian's statement. I originally started writing scripts in JS because I was more familar with it. I ran into several problem with scalability and threading issue that prompted me to look at LUA. > > LUA took less than a day to get a basic understand and an additional few hours to understand their version of objects. I had migrated all 15 of my JS scripts into functioning LUA script within 5 hours. Including re-creating the object-oriented nature of my JS source. > > LUA consumes far less resources and begins execution faster than JS. The only thing I need JS for at this time is cURL-like calls, as the built-in LUA has no equivalent to the JS Socket object. Having said that, the built in LUA can take advantage of LUA libraries that are installed on the OS, and LUA socket libraries do exists, so it is just a matter of me getting the time to install them. > > One more handy benefit for LUA, you can configure LUA scripts to run at switch startup to perform tasks (like a cron system) within the switch core. > > -pete > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] help with mod_conference stability > From: Brian West > Date: Thu, May 14, 2009 10:31 am > To: freeswitch-users at lists.freeswitch.org > > Its obvious if you look at the size of the JS VM vs the lua VM.. it would clearly scale better not needing megs and megs of ram per call vs about 160kb for lua. Lua took about 20 min to learn and about an hour to get the finer points down. > /b > > On May 14, 2009, at 12:24 PM, Stephen Crosby wrote: > > I keep hearing that lua is lighter weight / more scalable than javascript. I'd love to see some data that shows how big the difference really is. I could port all my scripts from javascript to lua, but it would require a lot of overhead (me learning lua + actually porting the scripts). I want to know how much benefit there really is to that. > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ________________________________ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/33e08c0c/attachment.html From lfurrea at gmail.com Thu May 14 13:45:01 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 14 May 2009 14:45:01 -0600 Subject: [Freeswitch-users] Help configuring external profile In-Reply-To: <004301c9d4d4$08f1f640$1ad5e2c0$@com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363808@cooper> <4FEB458A-2845-431E-A164-59CDD37AB4AD@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BB6363842@cooper> <013701c9d280$cbdf84c0$639e8e40$@com> <4F5E7E0C-D16E-4A59-97C3-83819EE02B2A@freeswitch.org> <014e01c9d287$27d5a2e0$7780e8a0$@com> <4F4597A9-B83C-4FF7-84AE-D97B6B46458D@freeswitch.org> <001101c9d4d1$a0914c60$e1b3e520$@com> <13B3E523-EC28-4E48-86D9-CBB30412FFA8@freeswitch.org> <004301c9d4d4$08f1f640$1ad5e2c0$@com> Message-ID: Lars, just start a new message and type in the address instead of editing an existing message to the list. On Thu, May 14, 2009 at 2:39 PM, Lars Zeb wrote: > Sorry about hijacking, but please tell me how I did that so I can avoid > it in the future. I merely sent a message to > freeswitch-users at lists.freeswitch.org with a subject of ?Help configuring > external profile?. If there?s another to post a question, I?d be glad to do > it. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Thursday, May 14, 2009 1:27 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Help configuring external profile > > > > Its guessing the IP you'll have to set it to the IP you want by opening up > the sofia profile in sip_profiles/internal.xml and external.xml and specify > the EXACT IP to use. > > > > On another note please stop hijacking threads. > > > > /b > > > > On May 14, 2009, at 3:21 PM, Lars Zeb wrote: > > > > How is FS getting 64.105.128.82 in ?sofia status? and how can I change > that to 64.105.128.83? > > Thanks, Lars > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/c7b1788d/attachment-0001.html From dave at 3c.co.uk Thu May 14 14:56:04 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 14 May 2009 22:56:04 +0100 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> Message-ID: <1242338164.4361.8.camel@dk-d820> On Thu, 2009-05-14 at 12:31 -0500, Brian West wrote: > Its obvious if you look at the size of the JS VM vs the lua VM.. it > would clearly scale better not needing megs and megs of ram per call > vs about 160kb for lua. Isn't this a bit of a non sequitur, given that there ought to be just one copy of the JS interpreter in RAM which is shared across all of the calls/threads which need to use it? --Dave From dftoro at yahoo.com Thu May 14 16:39:40 2009 From: dftoro at yahoo.com (Diego Toro) Date: Thu, 14 May 2009 16:39:40 -0700 (PDT) Subject: [Freeswitch-users] FS loses registration Message-ID: <826143.3303.qm@web33503.mail.mud.yahoo.com> I have a trouble, when FS start it register whit Siemens pbx, but after FS?loses registration.? I have a x-lite registered whit Siemens pbx and x-lite keeps register. ? I add whireshark trace? thank you ? Diego ------------------------------------------------------ whireshark trace? (register nok) ? ? ? VIA (x-lite): Via: SIP/2.0/UDP 192.168.0.125:27504;branch=z9hG4bK-d8754z-ed554d5f6c56a345-1---d8754z-;rport Transport: UDP Sent-by Address: 192.168.0.125 Sent-by port: 27504 Branch: z9hG4bK-d8754z-ed554d5f6c56a345-1---d8754z- RPort: rport Max-Forwards: 70 ? ? VIA (FS) Via: SIP/2.0/UDP 192.168.0.125:5080;rport;branch=z9hG4bKc2BX53H64t2ta Transport: UDP Sent-by Address: 192.168.0.125 Sent-by port: 5080 RPort: rport Branch: z9hG4bKc2BX53H64t2ta Max-Forwards: 70 ? ? To (x-lite) To: "135" SIP Display info: "135" SIP to address: sip:135 at 192.168.0.120 ? To (FS) To: SIP to address: sip:134 at 192.168.0.120 ? ? contact (x-lite) Contact: ?Contact Binding: ? ?URI: ? ? ?SIP contact address: sip:135 at 192.168.0.125:27504 ? contact (FS) Contact: ?Contact Binding: ? URI: ? ? SIP contact address: sip:gw+siemens3500 at 192.168.0.125:5080 Expires: 0 ? ? ? ============================= ? FS start up whireshark trace? (register ok) ? ? To: SIP to address: sip:134 at 192.168.0.120 Call-ID: 27c589ef-6e9a-5544-887a-52c0c7856977 ? ? Contact: ?Contact Binding: ? URI: ? ? SIP contact address: sip:gw+siemens3500 at 192.168.0.125:5080 Expires: 120 ? ? Via: SIP/2.0/UDP 192.168.0.125:5080;rport;branch=z9hG4bKBB461XN46BmcK Transport: UDP Sent-by Address: 192.168.0.125 Sent-by port: 5080 RPort: rport Branch: z9hG4bKBB461XN46BmcK Max-Forwards: 70 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/8ec46681/attachment.html From brian at freeswitch.org Thu May 14 16:49:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 14 May 2009 18:49:31 -0500 Subject: [Freeswitch-users] FS loses registration In-Reply-To: <826143.3303.qm@web33503.mail.mud.yahoo.com> References: <826143.3303.qm@web33503.mail.mud.yahoo.com> Message-ID: <3514E554-9D2A-44A1-8E1D-5B15A1ECBF4B@freeswitch.org> Seem to have left out a few things... /b On May 14, 2009, at 6:39 PM, Diego Toro wrote: > I have a trouble, when FS start it register whit Siemens pbx, but > after FS loses registration. I have a x-lite registered whit > Siemens pbx and x-lite keeps register. > > I add whireshark trace > > thank you > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090514/61382195/attachment.html From vijay11tiwari at hotmail.com Thu May 14 22:54:46 2009 From: vijay11tiwari at hotmail.com (vijay tiwari) Date: Fri, 15 May 2009 11:24:46 +0530 Subject: [Freeswitch-users] context is not working In-Reply-To: <7E62A325-D0BA-40CF-AF9D-66EA37A67813@avgs.ca> References: <7E62A325-D0BA-40CF-AF9D-66EA37A67813@avgs.ca> Message-ID: hello Math/Ray please find the user directory. and US context diaplan. [root at xxxx directory]# ls default default.xml US US.xml [root at xxxx directory]# cd US [root at xxxx US]# ls 12345.xml 8989.xml [root at xxx US]#vi 12345.xml [root at xxx US]#vi 8989.xml US context dailplan thanks vijay _________________________________________________________________ Live Search extreme As India feels the heat of poll season, get all the info you need on the MSN News Aggregator http://news.in.msn.com/National/indiaelections2009/aggregator/default.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/c7f0648b/attachment-0001.html From andy at fabulous4.co.uk Fri May 15 00:12:43 2009 From: andy at fabulous4.co.uk (Andy) Date: Fri, 15 May 2009 08:12:43 +0100 Subject: [Freeswitch-users] DTMF not comming through on some calls Message-ID: Hi, I have an urgent issue if anyone can help. I have been running freeswitch for 3-4 weeks now without issue. In the last 2 days some of the calls coming into the switch seem to get set up in such a way that means they cannot carry DTMF. ie on that call, no dtmf signals come through from the phone. It's not that digits get dropped some calls semm to handle dtmf perfectly and others don't seem to get dtmf at all. Can anyone shed any light opn this or suggest any solutions? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/fb57c551/attachment.html From jason at jasonjgw.net Fri May 15 00:47:08 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 15 May 2009 17:47:08 +1000 Subject: [Freeswitch-users] DTMF not comming through on some calls In-Reply-To: References: Message-ID: <20090515074708.GA8102@jdc.jasonjgw.net> Andy wrote: > It's not that digits get dropped some calls semm to handle dtmf perfectly > and others don't seem to get dtmf at all. > > Can anyone shed any light opn this or suggest any solutions? I can't help, but you could make it a lot easier for others to help you by including the necessary information with your question. For example, what DTMF method is configured in the SIP profiles - RFC2833 or Info, or are you using inband DTMF detection? What are the phones, and how are they connected to your FreeSWITCH system? What relevant information appears in your FreeSWITCH logs? For example, when debug-level logging is enabled, you should see log entries related to the DTMF detection. Check whether there are differences between the calls that work and those which don't. If it appears to be a bug, test whether you can reproduce it with the latest code taken from svn trunk. From andy at fabulous4.co.uk Fri May 15 02:41:24 2009 From: andy at fabulous4.co.uk (Andy) Date: Fri, 15 May 2009 10:41:24 +0100 Subject: [Freeswitch-users] DTMF not comming through on some calls In-Reply-To: <20090515074708.GA8102@jdc.jasonjgw.net> References: <20090515074708.GA8102@jdc.jasonjgw.net> Message-ID: <0D1A59E8ECBE45219C4925808342E454@D810> Apologies. The freeswitch software is receiving incoming calls from a voip gateway. I'm using voiptalk in the UK. The DTMF method was efault which I believe is "info" but I've now set it explicitly to rfc2833 inband to see if that helps. Is there a way I can tell from the logs that this is the case and that my config changes have worked. Most of the phones are mobiles but some landlines as well. I've done a detailed analysis of the logs and the calls that don't work are missing what appear to be critical actions in the debug. Namely: 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 And then a little later in the call.... 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp() Audio params changed for sofia/external/07540526194 at 194.145.190.143 from 194.145.190.143:11780 to 87.238.72.155:16968 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/07540526194 at 194.145.190.143] 77.86.49.249 port 21054 -> 87.238.72.155 port 16968 codec: 8 ms: 20 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968] 2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state() Processing Reinvite 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526194 at 194.145.190.143 entering state [completed][200] 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526194 at 194.145.190.143 entering state [ready][200] These lines appear for calls that work and not when they don't. Hope that helps. Cheers Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: 15 May 2009 08:47 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF not comming through on some calls Andy wrote: > It's not that digits get dropped some calls semm to handle dtmf > perfectly and others don't seem to get dtmf at all. > > Can anyone shed any light opn this or suggest any solutions? I can't help, but you could make it a lot easier for others to help you by including the necessary information with your question. For example, what DTMF method is configured in the SIP profiles - RFC2833 or Info, or are you using inband DTMF detection? What are the phones, and how are they connected to your FreeSWITCH system? What relevant information appears in your FreeSWITCH logs? For example, when debug-level logging is enabled, you should see log entries related to the DTMF detection. Check whether there are differences between the calls that work and those which don't. If it appears to be a bug, test whether you can reproduce it with the latest code taken from svn trunk. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jason at jasonjgw.net Fri May 15 03:19:42 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 15 May 2009 20:19:42 +1000 Subject: [Freeswitch-users] DTMF not comming through on some calls In-Reply-To: <0D1A59E8ECBE45219C4925808342E454@D810> References: <20090515074708.GA8102@jdc.jasonjgw.net> <0D1A59E8ECBE45219C4925808342E454@D810> Message-ID: <20090515101942.GA28363@jdc.jasonjgw.net> Andy wrote: > > The DTMF method was efault which I believe is "info" but I've now set it > explicitly to rfc2833 inband to see if that helps. Is there a way I can tell > from the logs that this is the case and that my config changes have worked. This is in the logs, and (assuming the logs you quoted were taken after any relevant configuration change), they indicate that RFC2833 is indeed being used. This is also the default in the supplied Sofia profiles. > > Most of the phones are mobiles but some landlines as well. > > I've done a detailed analysis of the logs and the calls that don't work are > missing what appear to be critical actions in the debug. Namely: > > 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set > 2833 dtmf payload to 101 That's turning on RFC2833, as I understand it, for DTMF detection. > > And then a little later in the call.... > > 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set > 2833 dtmf payload to 101 > 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp() > Audio params changed for sofia/external/07540526194 at 194.145.190.143 from > 194.145.190.143:11780 to 87.238.72.155:16968 > 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp() > AUDIO RTP [sofia/external/07540526194 at 194.145.190.143] 77.86.49.249 port > 21054 -> 87.238.72.155 port 16968 codec: 8 ms: 20 > 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() > AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968] > 2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state() > Processing Reinvite > 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel > sofia/external/07540526194 at 194.145.190.143 entering state [completed][200] > 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel > sofia/external/07540526194 at 194.145.190.143 entering state [ready][200] > > These lines appear for calls that work and not when they don't. There are obviously SIP reinvite messages being received from your SIP provider, and which FreeSWITCH is processing successfully. What I'm wondering is whether your provider is always offering RFC2833, since, given the above, they seem to have complex call handling arrangements. For that, you would need to look at the SDP from the remote end in the calls for which DTMF isn't being detected properly. Fortunately, this is logged by FreeSWITCH when set to debug logging. What you're looking for is a line such as a=rtpmap:101 telephone-event/8000 If that isn't present, then something odd would appear to be going on at your SIP provider's end, which is what I personally suspect, since FreeSWITCH is correctly activating RFC2833 support on the channel in other cases. You can also obtain a sip trace: sofia profile external siptrace on which will show you exactly what you're receiving from your provider. Disclaimer: I'm not an expert, but I hope this helps anyway. From yudha2008 at gmail.com Fri May 15 03:27:12 2009 From: yudha2008 at gmail.com (Baskar) Date: Fri, 15 May 2009 15:57:12 +0530 Subject: [Freeswitch-users] JavaScript session conference In-Reply-To: <87f2f3b90905140955y6758fdb5w61470affd43496e9@mail.gmail.com> References: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> <87f2f3b90905140955y6758fdb5w61470affd43496e9@mail.gmail.com> Message-ID: *Hi, Michael Collins Step1: I get Mobile Number and Extension Number from Database and pass those value to JavaScript. Step2: JavaScript will dial the both Mobile and Extension Number. After some time agent want to transfer the call to conference room. Step3: Then agent will dial another Mobile Number and transfer the call into same conference room. I want the call Flow to work in this sequence 9841799874==>1001 Transfer both the call in to conference Room 3001 == 9841799874 + 1002 Agent call another Mobile Number 9841799852 Transfer the call in to same Conference Room 3001=== 9841799852 + 9841799874 + 1002 all the 3 Number should be in conference 3001.* *Can you assist me to do this above process through JavaScript Thanks in advance ** -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/5160968a/attachment.html From dftoro at yahoo.com Fri May 15 05:58:20 2009 From: dftoro at yahoo.com (Diego Toro) Date: Fri, 15 May 2009 05:58:20 -0700 (PDT) Subject: [Freeswitch-users] FS loses registration Message-ID: <651698.82353.qm@web33506.mail.mud.yahoo.com> I have been changing gateway parameters but i don't know what is the problem. ? any suggestion ? ? Diego --- On Thu, 5/14/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users at lists.freeswitch.org Date: Thursday, May 14, 2009, 6:49 PM Seem to have left out a few things... /b On May 14, 2009, at 6:39 PM, Diego Toro wrote: I have a trouble, when FS start it register whit Siemens pbx, but after FS?loses registration.? I have a x-lite registered whit Siemens pbx and x-lite keeps register. ? I add whireshark trace? thank you ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/75fa6bb5/attachment-0001.html From anthony.minessale at gmail.com Fri May 15 06:09:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 May 2009 08:09:21 -0500 Subject: [Freeswitch-users] context is not working In-Reply-To: References: <7E62A325-D0BA-40CF-AF9D-66EA37A67813@avgs.ca> Message-ID: <191c3a030905150609o197f5aebr5669de38412c2769@mail.gmail.com> you don't really explain the paths of any of the files there. is US.xml in conf/dialplan because that is the only place it would get included from. look at log/freeswitch.xml.fsxml to see the fully expanded document and look for your US it belongs in the section not The dialplan section comes default with a public and default context to demonstrate how it works. You should rephrase things like "it's not working" to "I am not doing it right" until you get the hang of things...... On Fri, May 15, 2009 at 12:54 AM, vijay tiwari wrote: > > hello Math/Ray > > please find the user directory. and US context diaplan. > > > [root at xxxx directory]# ls > default default.xml US US.xml > [root at xxxx directory]# cd US > [root at xxxx US]# ls > 12345.xml 8989.xml > [root at xxx US]#vi 12345.xml > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > [root at xxx US]#vi 8989.xml > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > > > US context dailplan > > > > > > > > > expression="^(911|411|611|711)$" break="never"> > data="sofia/gateway/cordia/$1"/> > > expression="^([2-9][0-9]{2}[2-9][0-9]{6})$" break="never"> > data="sofia/gateway/cordia/1$1"/> > > expression="^(1[2-9][0-9]{2}[2-9][0-9]{6})$" break="never"> > data="sofia/gateway/cordia/$1"/> > > > data="sofia/gateway/cordia/$1"/> > > > > > > > thanks > vijay > > ------------------------------ > Get easy photo sharing with Windows LiveT Photos. Drag n' drop > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/eb1da396/attachment.html From anthony.minessale at gmail.com Fri May 15 06:20:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 May 2009 08:20:39 -0500 Subject: [Freeswitch-users] help with mod_conference stability In-Reply-To: <1242338164.4361.8.camel@dk-d820> References: <11990ade0905042212i68a94621ofe222128e7c72306@mail.gmail.com> <37B2E7B4-7379-405A-B90E-BC45828D0093@freeswitch.org> <11990ade0905050742r2b87bf99s6695e3c0b0f2e676@mail.gmail.com> <11990ade0905050748v39d1565cu54ea746b4f71e243@mail.gmail.com> <191c3a030905050825r397adc90ia4d517d852246d17@mail.gmail.com> <93cdabd20905140932ya0f36fbmd76feca79924a972@mail.gmail.com> <11990ade0905141024x6cddabdbie0c49aea5d11656f@mail.gmail.com> <1242338164.4361.8.camel@dk-d820> Message-ID: <191c3a030905150620kfc0cd3bi73f910f5eb29d229@mail.gmail.com> Unfortunately, no scripting language we have encountered has ever done proper threading to the point that you could even begin to do that. Lua is one of the better ones in a threaded env only because they solved the problem by making a tiny VM that can be easily reproduced thousands of times. You still can't share a lua vm across threads or it will crash because they use lots of setjmp longjmp calls to travel between their virtual stack and the real one. As a matter of fact, spidermonkey has code that tries to be threadsafe (but you have to enable it, --with-threadsafe makes me nervous =]) and it shares as much as it can already and we have debated changing it so it creates a whole JS instance per call just because of all the garbage collection and thread safety issues we have encountered. you have to call a whole bunch of hidden mutexes marked suspend and resume etc everywhere in the code where you transition from executing script side code to a blocking FreeSWITCH function. Really, the only point in my original comment was that if you want to scale up something that uses scripting the first way to save resources is to not leave the script open while you do something you know will take a long time "playing a 2 hour file, bridging a call, joining a conference" because the script resources are allocated and doing nothing this whole time and they all share a global garbage collector. if you don't need to remain in the script just transfer the call to a static extension and close it. It's just a suggestion really. On Thu, May 14, 2009 at 4:56 PM, David Knell wrote: > On Thu, 2009-05-14 at 12:31 -0500, Brian West wrote: > > Its obvious if you look at the size of the JS VM vs the lua VM.. it > > would clearly scale better not needing megs and megs of ram per call > > vs about 160kb for lua. > > Isn't this a bit of a non sequitur, given that there ought to be just > one copy of the JS interpreter in RAM which is shared across all of the > calls/threads which need to use it? > > --Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/cb942bf3/attachment.html From anthony.minessale at gmail.com Fri May 15 06:23:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 15 May 2009 08:23:16 -0500 Subject: [Freeswitch-users] FS loses registration In-Reply-To: <651698.82353.qm@web33506.mail.mud.yahoo.com> References: <651698.82353.qm@web33506.mail.mud.yahoo.com> Message-ID: <191c3a030905150623l2ab7e4dao26e8ee664a633555@mail.gmail.com> tport=tcp;transport=udp ??? reduce the register frequency to a smaller number like 30 seconds. On Fri, May 15, 2009 at 7:58 AM, Diego Toro wrote: > I have been changing gateway parameters but i don't know what is the > problem. > > any suggestion ? > > Diego > > --- On *Thu, 5/14/09, Brian West * wrote: > > > From: Brian West > Subject: Re: [Freeswitch-users] FS loses registration > To: freeswitch-users at lists.freeswitch.org > Date: Thursday, May 14, 2009, 6:49 PM > > > Seem to have left out a few things... > /b > > On May 14, 2009, at 6:39 PM, Diego Toro wrote: > > I have a trouble, when FS start it register whit Siemens pbx, but after > FS loses registration. I have a x-lite registered whit Siemens pbx and > x-lite keeps register. > > I add whireshark trace > > thank you > > Diego > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/fad7a975/attachment-0001.html From dujinfang at gmail.com Fri May 15 07:44:34 2009 From: dujinfang at gmail.com (dujinfang) Date: Fri, 15 May 2009 22:44:34 +0800 Subject: [Freeswitch-users] rtp stat marked all packets as cng on B-leg from xml_cdr Message-ID: <12A72C15-140F-4E40-B982-E413ADDE660F@gmail.com> I'm collecting rtp stat by using xml_cdr, weird that the b leg marked all packets as cng. Is that a problem? Here is a real call last more than 15 minutes (codec PCMU). FreeSWITCH is in the middle, A leg is another FS server and B leg is a PolyLink Sip Server. FreeSWITCH version is 13066. A-leg: 7718374 7712484 44878 44841 15560 0 0 3 3 7860786 7860744 45705 45702 0 0 3 B-leg: 7861260 7861260 45705 45705 481 0 0 0 0 7703020 0 44785 0 0 0 44785 Thanks. From dftoro at yahoo.com Fri May 15 07:57:19 2009 From: dftoro at yahoo.com (Diego Toro) Date: Fri, 15 May 2009 07:57:19 -0700 (PDT) Subject: [Freeswitch-users] FS loses registration Message-ID: <801060.89314.qm@web33507.mail.mud.yahoo.com> Hi, thanks your response ? tport=tcp is a contact-params value, i cut it now, but the problem continue, ? I checked freeswitch-user list and i saw a?entry "Double Re-Register problem"? http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg11559.html?i have the?same situation ? there are something that i can? do ? ? Diego --- On Fri, 5/15/09, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users at lists.freeswitch.org Date: Friday, May 15, 2009, 8:23 AM tport=tcp;transport=udp ??? reduce the register frequency to a smaller number like 30 seconds. On Fri, May 15, 2009 at 7:58 AM, Diego Toro wrote: I have been changing gateway parameters but i don't know what is the problem. ? any suggestion ? ? Diego --- On Thu, 5/14/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users at lists.freeswitch.org Date: Thursday, May 14, 2009, 6:49 PM Seem to have left out a few things... /b On May 14, 2009, at 6:39 PM, Diego Toro wrote: I have a trouble, when FS start it register whit Siemens pbx, but after FS?loses registration.? I have a x-lite registered whit Siemens pbx and x-lite keeps register. ? I add whireshark trace? thank you ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/b98e2804/attachment.html From bwood-lists at jacosoft.com Fri May 15 09:33:30 2009 From: bwood-lists at jacosoft.com (Brian Wood) Date: Fri, 15 May 2009 11:33:30 -0500 Subject: [Freeswitch-users] FXS ports not working on Sangoma A200 Message-ID: <4A0D995A.7060505@jacosoft.com> I have a Sangoma A200 with 1 FXO and 1 FXS module. Previously, I was using zaptel under the wanpipe-3.2.7 drivers. It worked fine in this configuration, but DTMF recognition was a bit flakey. I am trying to switch over to the native wanpipe/TDM interface with their 3.4.1 drivers and FreeSWITCH trunk (r13288 was my last update). The FXO ports are working fine. Hardware DTMF recognition works, which is great. The FXS ports are weird. When I pickup an analog phone, there is no dial tone on the FXS ports, just silence. The tones file is in place in /etc/openzap and /usr/local/freeswitch/conf (and I do see FreeSWITCH grabbing them during startup). I can originate a call to them, and they ring, but FreeSWITCH says there is no answer, and no audio comes across: 2009-05-15 16:26:28 [DEBUG] sofia.c:4331 sofia_handle_sip_i_invite() IP 192.168.98.143 Rejected by acl "domains". Falling back to Digest auth. 2009-05-15 16:26:28 [DEBUG] sofia.c:4331 sofia_handle_sip_i_invite() IP 192.168.98.143 Rejected by acl "domains". Falling back to Digest auth. 2009-05-15 16:26:28 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/1001 at 192.168.98.1 [234f57aa-9dbf-4ca8-aeca-a4b89c7cdb1e] 2009-05-15 16:26:28 [DEBUG] sofia.c:4958 sofia_handle_sip_i_invite() Setting NAT mode based on rfc1918 2009-05-15 16:26:28 [DEBUG] sofia.c:3001 sofia_handle_sip_i_state() Channel sofia/internal/1001 at 192.168.98.1 entering state [received][100] 2009-05-15 16:26:28 [DEBUG] sofia.c:3008 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 1242404734 1242404734 IN IP4 192.168.98.143 s=Polycom IP Phone c=IN IP4 192.168.98.143 t=0 0 a=sendrecv m=audio 2254 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:2939 sofia_glue_negotiate_sdp() Audio Codec Compare [G722:9:8000:0]/[G722:9:8000:20] 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:1912 sofia_glue_tech_set_codec() Set Codec sofia/internal/1001 at 192.168.98.1 G722/8000 20 ms 160 samples 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:2899 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running State Change CS_NEW 2009-05-15 16:26:28 [DEBUG] sofia.c:3167 sofia_handle_sip_i_state() (sofia/internal/1001 at 192.168.98.1) State Change CS_NEW -> CS_INIT 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 192.168.98.1 [BREAK] 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:403 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State NEW 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running State Change CS_INIT 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State INIT 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at 192.168.98.1 SOFIA INIT 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at 192.168.98.1) State Change CS_INIT -> CS_ROUTING 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 192.168.98.1 [BREAK] 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State INIT going to sleep 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running State Change CS_ROUTING 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State ROUTING 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at 192.168.98.1 SOFIA ROUTING 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/1001 at 192.168.98.1 Standard ROUTING 2009-05-15 16:26:28 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 1001->6001 in context default Dialplan: sofia/internal/1001 at 192.168.98.1 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001 at 192.168.98.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.98.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.98.1 parsing [default->analog-phone] continue=false Dialplan: sofia/internal/1001 at 192.168.98.1 Regex (PASS) [analog-phone] destination_number(6001) =~ /^6001$/ break=on-false Dialplan: sofia/internal/1001 at 192.168.98.1 Action answer() Dialplan: sofia/internal/1001 at 192.168.98.1 Action set(call_timeout=30) Dialplan: sofia/internal/1001 at 192.168.98.1 Action bridge(OpenZAP/2/1/1) 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1001 at 192.168.98.1) State Change CS_ROUTING -> CS_EXECUTE 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 192.168.98.1 [BREAK] 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State ROUTING going to sleep 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running State Change CS_EXECUTE 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State EXECUTE 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1001 at 192.168.98.1 SOFIA EXECUTE 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1001 at 192.168.98.1 Standard EXECUTE EXECUTE sofia/internal/1001 at 192.168.98.1 answer() 2009-05-15 16:26:28 [DEBUG] mod_dptools.c:649 answer_function() sofia/internal/1001 at 192.168.98.1 receive message [ANSWER] 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/1001 at 192.168.98.1] 192.168.98.1 port 31952 -> 192.168.98.143 port 2254 codec: 9 ms: 20 2009-05-15 16:26:28 [DEBUG] switch_rtp.c:906 switch_rtp_create() Starting timer [soft] 160 bytes per 20ms 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:534 sofia_answer_channel() Local SDP sofia/internal/1001 at 192.168.98.1: v=0 o=FreeSWITCH 1242372836 1242372837 IN IP4 192.168.98.1 s=FreeSWITCH c=IN IP4 192.168.98.1 t=0 0 m=audio 31952 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/internal/1001 at 192.168.98.1 [BREAK] 2009-05-15 16:26:28 [NOTICE] mod_dptools.c:649 answer_function() Channel [sofia/internal/1001 at 192.168.98.1] has been answered 2009-05-15 16:26:28 [DEBUG] switch_channel.c:182 switch_channel_audio_sync() sofia/internal/1001 at 192.168.98.1 receive message [AUDIO_SYNC] EXECUTE sofia/internal/1001 at 192.168.98.1 set(call_timeout=30) 2009-05-15 16:26:28 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1001 at 192.168.98.1 SET [call_timeout]=[30] 2009-05-15 16:26:28 [DEBUG] sofia.c:3001 sofia_handle_sip_i_state() Channel sofia/internal/1001 at 192.168.98.1 entering state [completed][200] EXECUTE sofia/internal/1001 at 192.168.98.1 bridge(OpenZAP/2/1/1) 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:343 tech_init() Set codec PCMU 20ms 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:1060 channel_outgoing_channel() Connect outbound channel OpenZAP/2:1/1 2009-05-15 16:26:28 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel OpenZAP/2:1/1 [a2aa1a26-ec30-4d7c-9d8e-477521231fad] 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:1072 channel_outgoing_channel() (OpenZAP/2:1/1) State Change CS_NEW -> CS_INIT 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 [BREAK] 2009-05-15 16:26:28 [DEBUG] ozmod_analog.c:66 analog_fxs_outgoing_call() Changing state on 2:1 from DOWN to GENRING 2009-05-15 16:26:28 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run() ANALOG CHANNEL thread starting. 2009-05-15 16:26:28 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for GENRING 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:1262 on_fxs_signal() got FXS sig [PROGRESS] 2009-05-15 16:26:28 [NOTICE] mod_openzap.c:1278 on_fxs_signal() Ring-Ready OpenZAP/2:1/1! 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/2:1/1) Running State Change CS_INIT 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (OpenZAP/2:1/1) State INIT 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:367 channel_on_init() (OpenZAP/2:1/1) State Change CS_INIT -> CS_ROUTING 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 [BREAK] 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (OpenZAP/2:1/1) State INIT going to sleep 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/2:1/1) Running State Change CS_ROUTING 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (OpenZAP/2:1/1) State ROUTING 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:390 channel_on_routing() OpenZAP/2:1/1 CHANNEL ROUTING 2009-05-15 16:26:28 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (OpenZAP/2:1/1) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 [BREAK] 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (OpenZAP/2:1/1) State ROUTING going to sleep 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/2:1/1) Running State Change CS_CONSUME_MEDIA 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (OpenZAP/2:1/1) State CONSUME_MEDIA 2009-05-15 16:26:28 [DEBUG] sofia.c:3001 sofia_handle_sip_i_state() Channel sofia/internal/1001 at 192.168.98.1 entering state [ready][200] 2009-05-15 16:26:58 [NOTICE] switch_ivr_originate.c:1957 switch_ivr_originate() Hangup OpenZAP/2:1/1 [CS_CONSUME_MEDIA] [NO_ANSWER] 2009-05-15 16:26:58 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal OpenZAP/2:1/1 [KILL] 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 [BREAK] 2009-05-15 16:26:58 [DEBUG] switch_ivr_originate.c:2101 switch_ivr_originate() Originate Resulted in Error Cause: 19 [NO_ANSWER] 2009-05-15 16:26:58 [INFO] mod_dptools.c:2074 audio_bridge_function() Originate Failed. Cause: NO_ANSWER 2009-05-15 16:26:58 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1001 at 192.168.98.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-15 16:26:58 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/1001 at 192.168.98.1 [KILL] 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 192.168.98.1 [BREAK] 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State EXECUTE going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running State Change CS_HANGUP 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State HANGUP 2009-05-15 16:26:58 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/internal/1001 at 192.168.98.1 hanging up, cause: NORMAL_CLEARING 2009-05-15 16:26:58 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending BYE to sofia/internal/1001 at 192.168.98.1 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at 192.168.98.1 Standard HANGUP, cause: NORMAL_CLEARING 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State HANGUP going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State Change CS_HANGUP -> CS_REPORTING 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 192.168.98.1 [BREAK] 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running State Change CS_REPORTING 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 192.168.98.1) State REPORTING 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/1001 at 192.168.98.1 Standard REPORTING, cause: NORMAL_CLEARING 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 192.168.98.1) State REPORTING going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State Change CS_REPORTING -> CS_DESTROY 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 1 (sofia/internal/1001 at 192.168.98.1) Locked, Waiting on external entities 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 1 (sofia/internal/1001 at 192.168.98.1) Ended 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1001 at 192.168.98.1 [CS_DESTROY] 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 192.168.98.1) State DESTROY 2009-05-15 16:26:58 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/internal/1001 at 192.168.98.1 SOFIA DESTROY 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/1001 at 192.168.98.1 Standard DESTROY 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 192.168.98.1) State DESTROY going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (OpenZAP/2:1/1) State CONSUME_MEDIA going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/2:1/1) Running State Change CS_HANGUP 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (OpenZAP/2:1/1) State HANGUP 2009-05-15 16:26:58 [DEBUG] mod_openzap.c:460 channel_on_hangup() Changing state on 2:1 from GENRING to HANGUP 2009-05-15 16:26:58 [DEBUG] mod_openzap.c:485 channel_on_hangup() OpenZAP/2:1/1 CHANNEL HANGUP 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:1/1 Standard HANGUP, cause: NO_ANSWER 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (OpenZAP/2:1/1) State HANGUP going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (OpenZAP/2:1/1) State Change CS_HANGUP -> CS_REPORTING 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 [BREAK] 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/2:1/1) Running State Change CS_REPORTING 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (OpenZAP/2:1/1) State REPORTING 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() OpenZAP/2:1/1 Standard REPORTING, cause: NO_ANSWER 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (OpenZAP/2:1/1) State REPORTING going to sleep 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:1/1) State Change CS_REPORTING -> CS_DESTROY 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 2 (OpenZAP/2:1/1) Locked, Waiting on external entities 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 2 (OpenZAP/2:1/1) Ended 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel OpenZAP/2:1/1 [CS_DESTROY] 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (OpenZAP/2:1/1) State DESTROY 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() OpenZAP/2:1/1 Standard DESTROY 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (OpenZAP/2:1/1) State DESTROY going to sleep 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for HANGUP 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:351 zap_analog_channel_run() Changing state on 2:1 from HANGUP to DOWN 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run() Executing state handler on 2:1 for DOWN 2009-05-15 16:26:58 [DEBUG] mod_openzap.c:1262 on_fxs_signal() got FXS sig [STOP] 2009-05-15 16:26:58 [DEBUG] zap_io.c:1179 zap_channel_done() channel done 2:1 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:726 zap_analog_channel_run() ANALOG CHANNEL 2:1 thread ended. When I pick up the phone attached to the FXS port, there is what sounds like a burst of data (caller ID?) and a series of about 10 clicks which end when FreeSWITCH closes the channel. Is there something obvious I'm missing or should try? Or should I take this up with Sangoma? Any suggestions would be greatly appreciated. I have tried all sorts of slight modifications to the relevant configuration files, with no change in the FXS port behavior, but they currently look like: gentoo conf # cat openzap.conf [span wanpipe] name => OpenZAP number => 5000 fxo-channel => 1:1 number => 5000 fxo-channel => 1:2 [span wanpipe] name => OpenZAP number => 4165551111 fxs-channel => 1:3 number => 4165552222 fxs-channel => 1:4 gentoo conf # cat autoload_configs/openzap.conf.xml gentoo conf # wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info | ------------------------------- 1 . AFT-A200-SH : SLOT=9 : BUS=1 : IRQ=16 : CPU=A : PORT=PRI : HWEC=32 : V=11 Card Cnt: A200=1 gentoo conf # cat /etc/wanpipe/wanpipe1.conf #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Mon Jul 31 17:10:23 EDT 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. #================================================ # Sangoma Technologies Inc. #================================================ [devices] wanpipe1 = WAN_AFT_ANALOG, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 9 PCIBUS = 1 FE_MEDIA = FXO/FXS TDMV_LAW = MULAW TDMV_OPERMODE = FCC RM_BATTTHRESH = 3 RM_BATTDEBOUNCE = 16 FE_NETWORK_SYNC = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = YES [w1g1] ACTIVE_CH = ALL MTU = 8 TDMV_HWEC = YES freeswitch at gentoo> oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: analog chan_count: 2 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options 3way +OK span: 2 (span2) type: analog chan_count: 2 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none And here's what it looks like when FreeSWITCH starts: 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/modules.conf. 2009-05-15 16:17:19 [NOTICE] zap_io.c:2609 zap_global_init() Modules configured: 1 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/openzap.conf. 2009-05-15 16:17:19 [DEBUG] zap_io.c:2213 load_config() found config for span 2009-05-15 16:17:19 [INFO] zap_io.c:2430 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/wanpipe.conf. 2009-05-15 16:17:19 [INFO] zap_io.c:2230 load_config() auto-loaded 'wanpipe' 2009-05-15 16:17:19 [DEBUG] zap_io.c:2251 load_config() created span 1 (span1) of type wanpipe 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 [name]=[OpenZAP] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 [number]=[5000] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 [fxo-channel]=[1:1] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2293 load_config() setting trunk type to 'FXO' start(KEWL) 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() configuring device s1c1 as OpenZAP device 1:1 fd:37 DTMF: hardware 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 [number]=[5000] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 [fxo-channel]=[1:2] 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() configuring device s1c2 as OpenZAP device 1:2 fd:38 DTMF: hardware 2009-05-15 16:17:19 [DEBUG] zap_io.c:2213 load_config() found config for span 2009-05-15 16:17:19 [DEBUG] zap_io.c:2251 load_config() created span 2 (span2) of type wanpipe 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 [name]=[OpenZAP] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 [number]=[4165551111] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 [fxs-channel]=[1:3] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2304 load_config() setting trunk type to 'FXS' start(KEWL) 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() configuring device s1c3 as OpenZAP device 2:1 fd:39 DTMF: hardware 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 [number]=[4165552222] 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 [fxs-channel]=[1:4] 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() configuring device s1c4 as OpenZAP device 2:2 fd:40 DTMF: hardware 2009-05-15 16:17:19 [INFO] zap_io.c:2353 load_config() Configured 4 channel(s) 2009-05-15 16:17:19 [INFO] zap_io.c:2447 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so 2009-05-15 16:17:19 [INFO] zap_io.c:2563 zap_configure_span() auto-loaded 'analog' 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/tones.conf. 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [dial] = [v=-7;%(1000,0,350,440)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [dial] = [350,440] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [ring] = [v=-7;%(2000,4000,440,480)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [ring] = [440,480] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [busy] = [v=-7;%(500,500,480,620)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [busy] = [480,620] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [attn] = [1400,2060,2450,2600] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [callwaiting-sas] = [v=0;%(300,0,440)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [callwaiting-sas] = [440] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [callwaiting-cas] = [2750,2130] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [fail1] = [913.8] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [fail2] = [1370.6] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [fail3] = [776.7] 2009-05-15 16:17:19 [DEBUG] ozmod_analog.c:875 zap_analog_run() ANALOG thread starting. 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() Configuration file is /usr/local/freeswitch/conf/tones.conf. 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [dial] = [v=-7;%(1000,0,350,440)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [dial] = [350,440] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [ring] = [v=-7;%(2000,4000,440,480)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [ring] = [440,480] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [busy] = [v=-7;%(500,500,480,620)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [busy] = [480,620] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [attn] = [1400,2060,2450,2600] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [callwaiting-sas] = [v=0;%(300,0,440)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [callwaiting-sas] = [440] 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [callwaiting-cas] = [2750,2130] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [fail1] = [913.8] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [fail2] = [1370.6] 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added tone detect [fail3] = [776.7] 2009-05-15 16:17:19 [DEBUG] ozmod_analog.c:875 zap_analog_run() ANALOG thread starting. 2009-05-15 16:17:19 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-05-15 16:17:19 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-05-15 16:17:19 [NOTICE] switch_loadable_module.c:248 switch_loadable_module_process() Adding Application 'disable_ec' 2009-05-15 16:17:19 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'oz' From mike at jerris.com Fri May 15 09:58:49 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 15 May 2009 12:58:49 -0400 Subject: [Freeswitch-users] FXS ports not working on Sangoma A200 In-Reply-To: <4A0D995A.7060505@jacosoft.com> References: <4A0D995A.7060505@jacosoft.com> Message-ID: <72EC19D5-0359-4A34-A1C7-15557BA41921@jerris.com> try the 3.5.2 drivers. Mike On May 15, 2009, at 12:33 PM, Brian Wood wrote: > I have a Sangoma A200 with 1 FXO and 1 FXS module. Previously, I was > using zaptel under the wanpipe-3.2.7 drivers. It worked fine in this > configuration, but DTMF recognition was a bit flakey. > > I am trying to switch over to the native wanpipe/TDM interface with > their 3.4.1 drivers and FreeSWITCH trunk (r13288 was my last update). > > The FXO ports are working fine. Hardware DTMF recognition works, which > is great. > > The FXS ports are weird. > > When I pickup an analog phone, there is no dial tone on the FXS ports, > just silence. The tones file is in place in /etc/openzap and > /usr/local/freeswitch/conf (and I do see FreeSWITCH grabbing them > during > startup). > > I can originate a call to them, and they ring, but FreeSWITCH says > there > is no answer, and no audio comes across: > > 2009-05-15 16:26:28 [DEBUG] sofia.c:4331 sofia_handle_sip_i_invite() > IP > 192.168.98.143 Rejected by acl "domains". Falling back to Digest auth. > 2009-05-15 16:26:28 [DEBUG] sofia.c:4331 sofia_handle_sip_i_invite() > IP > 192.168.98.143 Rejected by acl "domains". Falling back to Digest auth. > 2009-05-15 16:26:28 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel sofia/internal/1001 at 192.168.98.1 > [234f57aa-9dbf-4ca8-aeca-a4b89c7cdb1e] > 2009-05-15 16:26:28 [DEBUG] sofia.c:4958 sofia_handle_sip_i_invite() > Setting NAT mode based on rfc1918 > 2009-05-15 16:26:28 [DEBUG] sofia.c:3001 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at 192.168.98.1 entering state [received] > [100] > 2009-05-15 16:26:28 [DEBUG] sofia.c:3008 sofia_handle_sip_i_state() > Remote SDP: > v=0 > o=- 1242404734 1242404734 IN IP4 192.168.98.143 > s=Polycom IP Phone > c=IN IP4 192.168.98.143 > t=0 0 > a=sendrecv > m=audio 2254 RTP/AVP 9 0 8 18 101 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > > 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:2939 > sofia_glue_negotiate_sdp() > Audio Codec Compare [G722:9:8000:0]/[G722:9:8000:20] > 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:1912 > sofia_glue_tech_set_codec() Set Codec sofia/internal/1001 at 192.168.98.1 > G722/8000 20 ms 160 samples > 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:2899 > sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running > State Change CS_NEW > 2009-05-15 16:26:28 [DEBUG] sofia.c:3167 sofia_handle_sip_i_state() > (sofia/internal/1001 at 192.168.98.1) State Change CS_NEW -> CS_INIT > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 192.168.98.1 [BREAK] > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:403 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State NEW > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running > State Change CS_INIT > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > INIT > 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/internal/1001 at 192.168.98.1 SOFIA INIT > 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/internal/1001 at 192.168.98.1) State Change CS_INIT -> CS_ROUTING > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 192.168.98.1 [BREAK] > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > INIT > going to sleep > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running > State Change CS_ROUTING > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > ROUTING > 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/1001 at 192.168.98.1 SOFIA ROUTING > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:78 > switch_core_standard_on_routing() sofia/internal/1001 at 192.168.98.1 > Standard ROUTING > 2009-05-15 16:26:28 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 1001->6001 in context default > Dialplan: sofia/internal/1001 at 192.168.98.1 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/1001 at 192.168.98.1 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1001 at 192.168.98.1 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1001 at 192.168.98.1 parsing > [default->analog-phone] continue=false > Dialplan: sofia/internal/1001 at 192.168.98.1 Regex (PASS) [analog-phone] > destination_number(6001) =~ /^6001$/ break=on-false > Dialplan: sofia/internal/1001 at 192.168.98.1 Action answer() > Dialplan: sofia/internal/1001 at 192.168.98.1 Action set(call_timeout=30) > Dialplan: sofia/internal/1001 at 192.168.98.1 Action bridge(OpenZAP/ > 2/1/1) > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:114 > switch_core_standard_on_routing() (sofia/internal/1001 at 192.168.98.1) > State Change CS_ROUTING -> CS_EXECUTE > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 192.168.98.1 [BREAK] > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > ROUTING going to sleep > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running > State Change CS_EXECUTE > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > EXECUTE > 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:173 sofia_on_execute() > sofia/internal/1001 at 192.168.98.1 SOFIA EXECUTE > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:151 > switch_core_standard_on_execute() sofia/internal/1001 at 192.168.98.1 > Standard EXECUTE > EXECUTE sofia/internal/1001 at 192.168.98.1 answer() > 2009-05-15 16:26:28 [DEBUG] mod_dptools.c:649 answer_function() > sofia/internal/1001 at 192.168.98.1 receive message [ANSWER] > 2009-05-15 16:26:28 [DEBUG] sofia_glue.c:2146 > sofia_glue_activate_rtp() > AUDIO RTP [sofia/internal/1001 at 192.168.98.1] 192.168.98.1 port 31952 > -> > 192.168.98.143 port 2254 codec: 9 ms: 20 > 2009-05-15 16:26:28 [DEBUG] switch_rtp.c:906 switch_rtp_create() > Starting timer [soft] 160 bytes per 20ms > 2009-05-15 16:26:28 [DEBUG] mod_sofia.c:534 sofia_answer_channel() > Local > SDP sofia/internal/1001 at 192.168.98.1: > v=0 > o=FreeSWITCH 1242372836 1242372837 IN IP4 192.168.98.1 > s=FreeSWITCH > c=IN IP4 192.168.98.1 > t=0 0 > m=audio 31952 RTP/AVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:630 > switch_core_session_perform_receive_message() Send signal > sofia/internal/1001 at 192.168.98.1 [BREAK] > 2009-05-15 16:26:28 [NOTICE] mod_dptools.c:649 answer_function() > Channel > [sofia/internal/1001 at 192.168.98.1] has been answered > 2009-05-15 16:26:28 [DEBUG] switch_channel.c:182 > switch_channel_audio_sync() sofia/internal/1001 at 192.168.98.1 receive > message [AUDIO_SYNC] > EXECUTE sofia/internal/1001 at 192.168.98.1 set(call_timeout=30) > 2009-05-15 16:26:28 [DEBUG] mod_dptools.c:748 set_function() > sofia/internal/1001 at 192.168.98.1 SET [call_timeout]=[30] > 2009-05-15 16:26:28 [DEBUG] sofia.c:3001 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at 192.168.98.1 entering state [completed] > [200] > EXECUTE sofia/internal/1001 at 192.168.98.1 bridge(OpenZAP/2/1/1) > 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:343 tech_init() Set codec > PCMU > 20ms > 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:1060 > channel_outgoing_channel() Connect outbound channel OpenZAP/2:1/1 > 2009-05-15 16:26:28 [NOTICE] switch_channel.c:602 > switch_channel_set_name() New Channel OpenZAP/2:1/1 > [a2aa1a26-ec30-4d7c-9d8e-477521231fad] > 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:1072 > channel_outgoing_channel() (OpenZAP/2:1/1) State Change CS_NEW -> > CS_INIT > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 > [BREAK] > 2009-05-15 16:26:28 [DEBUG] ozmod_analog.c:66 > analog_fxs_outgoing_call() > Changing state on 2:1 from DOWN to GENRING > 2009-05-15 16:26:28 [DEBUG] ozmod_analog.c:239 > zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2009-05-15 16:26:28 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > Executing state handler on 2:1 for GENRING > 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:1262 on_fxs_signal() got FXS > sig [PROGRESS] > 2009-05-15 16:26:28 [NOTICE] mod_openzap.c:1278 on_fxs_signal() > Ring-Ready OpenZAP/2:1/1! > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OpenZAP/2:1/1) Running State Change CS_INIT > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (OpenZAP/2:1/1) State INIT > 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:367 channel_on_init() > (OpenZAP/2:1/1) State Change CS_INIT -> CS_ROUTING > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 > [BREAK] > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (OpenZAP/2:1/1) State INIT going to sleep > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OpenZAP/2:1/1) Running State Change > CS_ROUTING > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (OpenZAP/2:1/1) State ROUTING > 2009-05-15 16:26:28 [DEBUG] mod_openzap.c:390 channel_on_routing() > OpenZAP/2:1/1 CHANNEL ROUTING > 2009-05-15 16:26:28 [DEBUG] switch_ivr_originate.c:63 > originate_on_routing() (OpenZAP/2:1/1) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-05-15 16:26:28 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 > [BREAK] > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (OpenZAP/2:1/1) State ROUTING going to sleep > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OpenZAP/2:1/1) Running State Change > CS_CONSUME_MEDIA > 2009-05-15 16:26:28 [DEBUG] switch_core_state_machine.c:502 > switch_core_session_run() (OpenZAP/2:1/1) State CONSUME_MEDIA > 2009-05-15 16:26:28 [DEBUG] sofia.c:3001 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at 192.168.98.1 entering state [ready][200] > 2009-05-15 16:26:58 [NOTICE] switch_ivr_originate.c:1957 > switch_ivr_originate() Hangup OpenZAP/2:1/1 [CS_CONSUME_MEDIA] > [NO_ANSWER] > 2009-05-15 16:26:58 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal OpenZAP/2:1/1 [KILL] > 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 > [BREAK] > 2009-05-15 16:26:58 [DEBUG] switch_ivr_originate.c:2101 > switch_ivr_originate() Originate Resulted in Error Cause: 19 > [NO_ANSWER] > 2009-05-15 16:26:58 [INFO] mod_dptools.c:2074 audio_bridge_function() > Originate Failed. Cause: NO_ANSWER > 2009-05-15 16:26:58 [NOTICE] switch_core_state_machine.c:179 > switch_core_standard_on_execute() Hangup > sofia/internal/1001 at 192.168.98.1 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-05-15 16:26:58 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/internal/1001 at 192.168.98.1 [KILL] > 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 192.168.98.1 [BREAK] > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > EXECUTE going to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running > State Change CS_HANGUP > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > HANGUP > 2009-05-15 16:26:58 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/internal/1001 at 192.168.98.1 hanging up, cause: NORMAL_CLEARING > 2009-05-15 16:26:58 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending > BYE to sofia/internal/1001 at 192.168.98.1 > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/1001 at 192.168.98.1 > Standard HANGUP, cause: NORMAL_CLEARING > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > HANGUP going to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > Change CS_HANGUP -> CS_REPORTING > 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 192.168.98.1 [BREAK] > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) Running > State Change CS_REPORTING > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/internal/1001 at 192.168.98.1 > ) > State REPORTING > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/internal/1001 at 192.168.98.1 > Standard REPORTING, cause: NORMAL_CLEARING > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/internal/1001 at 192.168.98.1 > ) > State REPORTING going to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/internal/1001 at 192.168.98.1) State > Change CS_REPORTING -> CS_DESTROY > 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 1 > (sofia/internal/1001 at 192.168.98.1) Locked, Waiting on external > entities > 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 1 > (sofia/internal/1001 at 192.168.98.1) Ended > 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel > sofia/internal/1001 at 192.168.98.1 [CS_DESTROY] > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/internal/1001 at 192.168.98.1) > State DESTROY > 2009-05-15 16:26:58 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/internal/1001 at 192.168.98.1 SOFIA DESTROY > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/internal/1001 at 192.168.98.1 > Standard DESTROY > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/internal/1001 at 192.168.98.1) > State DESTROY going to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:502 > switch_core_session_run() (OpenZAP/2:1/1) State CONSUME_MEDIA going > to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OpenZAP/2:1/1) Running State Change > CS_HANGUP > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (OpenZAP/2:1/1) State HANGUP > 2009-05-15 16:26:58 [DEBUG] mod_openzap.c:460 channel_on_hangup() > Changing state on 2:1 from GENRING to HANGUP > 2009-05-15 16:26:58 [DEBUG] mod_openzap.c:485 channel_on_hangup() > OpenZAP/2:1/1 CHANNEL HANGUP > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() OpenZAP/2:1/1 Standard HANGUP, cause: > NO_ANSWER > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (OpenZAP/2:1/1) State HANGUP going to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (OpenZAP/2:1/1) State Change CS_HANGUP -> > CS_REPORTING > 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal OpenZAP/2:1/1 > [BREAK] > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OpenZAP/2:1/1) Running State Change > CS_REPORTING > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (OpenZAP/2:1/1) State REPORTING > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() OpenZAP/2:1/1 Standard REPORTING, > cause: NO_ANSWER > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (OpenZAP/2:1/1) State REPORTING > going to sleep > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (OpenZAP/2:1/1) State Change CS_REPORTING -> > CS_DESTROY > 2009-05-15 16:26:58 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 2 (OpenZAP/2:1/1) Locked, Waiting > on external entities > 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 2 (OpenZAP/2:1/1) Ended > 2009-05-15 16:26:58 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel OpenZAP/2:1/1 [CS_DESTROY] > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (OpenZAP/2:1/1) State DESTROY > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() OpenZAP/2:1/1 Standard DESTROY > 2009-05-15 16:26:58 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (OpenZAP/2:1/1) State DESTROY > going > to sleep > 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > Executing state handler on 2:1 for HANGUP > 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:351 > zap_analog_channel_run() > Changing state on 2:1 from HANGUP to DOWN > 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:410 > zap_analog_channel_run() > Executing state handler on 2:1 for DOWN > 2009-05-15 16:26:58 [DEBUG] mod_openzap.c:1262 on_fxs_signal() got FXS > sig [STOP] > 2009-05-15 16:26:58 [DEBUG] zap_io.c:1179 zap_channel_done() channel > done 2:1 > 2009-05-15 16:26:58 [DEBUG] ozmod_analog.c:726 > zap_analog_channel_run() > ANALOG CHANNEL 2:1 thread ended. > > When I pick up the phone attached to the FXS port, there is what > sounds > like a burst of data (caller ID?) and a series of about 10 clicks > which > end when FreeSWITCH closes the channel. > > Is there something obvious I'm missing or should try? Or should I take > this up with Sangoma? Any suggestions would be greatly appreciated. > > I have tried all sorts of slight modifications to the relevant > configuration files, with no change in the FXS port behavior, but they > currently look like: > > gentoo conf # cat openzap.conf > [span wanpipe] > name => OpenZAP > number => 5000 > fxo-channel => 1:1 > number => 5000 > fxo-channel => 1:2 > > [span wanpipe] > name => OpenZAP > number => 4165551111 > fxs-channel => 1:3 > number => 4165552222 > fxs-channel => 1:4 > > gentoo conf # cat autoload_configs/openzap.conf.xml > > > > > > > > > > > > > > > > > > > > > > > > > gentoo conf # wanrouter hwprobe > > ------------------------------- > | Wanpipe Hardware Probe Info | > ------------------------------- > 1 . AFT-A200-SH : SLOT=9 : BUS=1 : IRQ=16 : CPU=A : PORT=PRI : > HWEC=32 : > V=11 > > Card Cnt: A200=1 > > gentoo conf # cat /etc/wanpipe/wanpipe1.conf > #================================================ > # WANPIPE1 Configuration File > #================================================ > # > # Date: Mon Jul 31 17:10:23 EDT 2006 > # > # Note: This file was generated automatically > # by /usr/local/sbin/setup-sangoma program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > #================================================ > # Sangoma Technologies Inc. > #================================================ > > [devices] > wanpipe1 = WAN_AFT_ANALOG, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 9 > PCIBUS = 1 > FE_MEDIA = FXO/FXS > TDMV_LAW = MULAW > TDMV_OPERMODE = FCC > RM_BATTTHRESH = 3 > RM_BATTDEBOUNCE = 16 > FE_NETWORK_SYNC = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_HW_DTMF = YES > TDMV_HW_FAX_DETECT = YES > > [w1g1] > ACTIVE_CH = ALL > MTU = 8 > TDMV_HWEC = YES > > freeswitch at gentoo> oz list > API CALL [oz(list)] output: > +OK > span: 1 (span1) > type: analog > chan_count: 2 > dialplan: XML > context: default > dial_regex: > fail_dial_regex: > hold_music: > analog_options 3way > +OK > span: 2 (span2) > type: analog > chan_count: 2 > dialplan: XML > context: default > dial_regex: > fail_dial_regex: > hold_music: > analog_options none > > > And here's what it looks like when FreeSWITCH starts: > > 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() > Configuration file is /usr/local/freeswitch/conf/modules.conf. > 2009-05-15 16:17:19 [NOTICE] zap_io.c:2609 zap_global_init() Modules > configured: 1 > 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() > Configuration file is /usr/local/freeswitch/conf/openzap.conf. > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2213 load_config() found config > for > span > 2009-05-15 16:17:19 [INFO] zap_io.c:2430 zap_load_module() Loading IO > from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] > 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() > Configuration file is /usr/local/freeswitch/conf/wanpipe.conf. > 2009-05-15 16:17:19 [INFO] zap_io.c:2230 load_config() auto-loaded > 'wanpipe' > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2251 load_config() created span 1 > (span1) of type wanpipe > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 > [name]=[OpenZAP] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 > [number]=[5000] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 > [fxo-channel]=[1:1] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2293 load_config() setting trunk > type to 'FXO' start(KEWL) > 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() > configuring device s1c1 as OpenZAP device 1:1 fd:37 DTMF: hardware > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 > [number]=[5000] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 1 > [fxo-channel]=[1:2] > 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() > configuring device s1c2 as OpenZAP device 1:2 fd:38 DTMF: hardware > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2213 load_config() found config > for > span > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2251 load_config() created span 2 > (span2) of type wanpipe > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 > [name]=[OpenZAP] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 > [number]=[4165551111] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 > [fxs-channel]=[1:3] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2304 load_config() setting trunk > type to 'FXS' start(KEWL) > 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() > configuring device s1c3 as OpenZAP device 2:1 fd:39 DTMF: hardware > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 > [number]=[4165552222] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:2264 load_config() span 2 > [fxs-channel]=[1:4] > 2009-05-15 16:17:19 [INFO] ozmod_wanpipe.c:207 wp_open_range() > configuring device s1c4 as OpenZAP device 2:2 fd:40 DTMF: hardware > 2009-05-15 16:17:19 [INFO] zap_io.c:2353 load_config() Configured 4 > channel(s) > 2009-05-15 16:17:19 [INFO] zap_io.c:2447 zap_load_module() Loading SIG > from /usr/local/freeswitch/mod/ozmod_analog.so > 2009-05-15 16:17:19 [INFO] zap_io.c:2563 zap_configure_span() > auto-loaded 'analog' > 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() > Configuration file is /usr/local/freeswitch/conf/tones.conf. > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [dial] = [v=-7;%(1000,0,350,440)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [dial] = [350,440] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [ring] = [v=-7;%(2000,4000,440,480)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [ring] = [440,480] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [busy] = [v=-7;%(500,500,480,620)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [busy] = [480,620] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [attn] = [1400,2060,2450,2600] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [callwaiting-sas] = [v=0;%(300,0,440)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [callwaiting-sas] = [440] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [callwaiting-cas] = [2750,2130] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [fail1] = [913.8] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [fail2] = [1370.6] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [fail3] = [776.7] > 2009-05-15 16:17:19 [DEBUG] ozmod_analog.c:875 zap_analog_run() ANALOG > thread starting. > 2009-05-15 16:17:19 [DEBUG] zap_config.c:56 zap_config_open_file() > Configuration file is /usr/local/freeswitch/conf/tones.conf. > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [dial] = [v=-7;%(1000,0,350,440)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [dial] = [350,440] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [ring] = [v=-7;%(2000,4000,440,480)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [ring] = [440,480] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [busy] = [v=-7;%(500,500,480,620)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [busy] = [480,620] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [attn] = [1400,2060,2450,2600] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [callwaiting-sas] = [v=0;%(300,0,440)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [callwaiting-sas] = [440] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:474 zap_span_load_tones() added > tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [callwaiting-cas] = [2750,2130] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [fail1] = [913.8] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [fail2] = [1370.6] > 2009-05-15 16:17:19 [DEBUG] zap_io.c:472 zap_span_load_tones() added > tone detect [fail3] = [776.7] > 2009-05-15 16:17:19 [DEBUG] ozmod_analog.c:875 zap_analog_run() ANALOG > thread starting. > 2009-05-15 16:17:19 [CONSOLE] switch_loadable_module.c:889 > switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > 2009-05-15 16:17:19 [NOTICE] switch_loadable_module.c:142 > switch_loadable_module_process() Adding Endpoint 'openzap' > 2009-05-15 16:17:19 [NOTICE] switch_loadable_module.c:248 > switch_loadable_module_process() Adding Application 'disable_ec' > 2009-05-15 16:17:19 [NOTICE] switch_loadable_module.c:270 > switch_loadable_module_process() Adding API Function 'oz' > > ____ From msc at freeswitch.org Fri May 15 10:22:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 15 May 2009 10:22:44 -0700 Subject: [Freeswitch-users] JavaScript session conference In-Reply-To: References: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> <87f2f3b90905140955y6758fdb5w61470affd43496e9@mail.gmail.com> Message-ID: <87f2f3b90905151022x6cd999eeuf23ac95c8d57f706@mail.gmail.com> On Fri, May 15, 2009 at 3:27 AM, Baskar wrote: > *Hi, > > Michael Collins > > Step1: I get Mobile Number and Extension Number from Database and pass > those value to JavaScript. > * Who or what needs to get the mobile number and extension from a database? What happens that leads to this process? > * > Step2: JavaScript will dial the both Mobile and Extension Number. After > some time agent want to transfer the call to conference room. > * This is the first time I've heard mention of an agent. Sorry if I missed it earlier. Who is the agent and what does he/she actually do? > * > Step3: Then agent will dial another Mobile Number and transfer the call > into same conference room. > * What kind of telephone device will the agent be using? Hard phone, softphone, etc.? > * > I want the call Flow to work in this sequence > > 9841799874==>1001* Who is extension 1001 in this scenario? > * > > Transfer both the call in to conference Room 3001 == 9841799874 + 1002 > * Who is extension 1002 in this scenario? > * > Agent call another Mobile Number > > 9841799852 > > Transfer the call in to same Conference Room 3001=== > 9841799852 + 9841799874 + 1002 all the 3 Number should be in > conference 3001.* > > *Can you assist me to do this above process through JavaScript > > Thanks in advance > ** > -- > Warm Regards, > N.Baskar > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/9a0c15c3/attachment.html From bwood-lists at jacosoft.com Fri May 15 13:06:08 2009 From: bwood-lists at jacosoft.com (Brian Wood) Date: Fri, 15 May 2009 15:06:08 -0500 Subject: [Freeswitch-users] FXS ports not working on Sangoma A200 [solved] In-Reply-To: <72EC19D5-0359-4A34-A1C7-15557BA41921@jerris.com> References: <4A0D995A.7060505@jacosoft.com> <72EC19D5-0359-4A34-A1C7-15557BA41921@jerris.com> Message-ID: <4A0DCB30.2030803@jacosoft.com> Ah! I was looking at the dates not the version numbers on their site. The beta is listed as 2008-xx-yy and the stable 2009-xx-yy, so I grabbed what I thought was the newer one. Once I rebuilt everything, FXO and FXS are both working. FYI: The last commit to ozmod_wanpipe.c broke it for Linux, but I expect someone will notice this out soon enough. I had to back down to the previous revision to compile: Index: ozmod_wanpipe.c =================================================================== --- ozmod_wanpipe.c (revision 733) +++ ozmod_wanpipe.c (revision 732) @@ -36,35 +36,10 @@ #include #endif #include "openzap.h" -#ifndef __WINDOWS__ #include #include ... -#endif #include "libsangoma.h" -#if defined(__WINDOWS__) -/*! Backward compatible defines - current code is all using the old names*/ -#define sangoma_open_tdmapi_span_chan sangoma_open_api_span_chan -#define sangoma_open_tdmapi_span sangoma_open_api_span -#define sangoma_open_tdmapi_ctrl sangoma_open_api_ctrl -#define sangoma_tdm_get_fe_status sangoma_get_fe_status -#define sangoma_socket_close sangoma_close -#define sangoma_tdm_get_hw_coding sangoma_get_hw_coding -#define sangoma_tdm_set_fe_status sangoma_set_fe_status -#define sangoma_tdm_get_link_status sangoma_get_link_status -#define sangoma_tdm_flush_bufs sangoma_flush_bufs -#define sangoma_tdm_cmd_exec sangoma_cmd_exec -#define sangoma_tdm_read_event sangoma_read_event -#define sangoma_readmsg_tdm sangoma_readmsg -#define sangoma_readmsg_socket sangoma_readmsg -#define sangoma_sendmsg_socket sangoma_writemsg -#define sangoma_writemsg_tdm sangoma_writemsg -#define sangoma_create_socket_intr sangoma_open_api_span_chan -#define EX_DECLARE_DATA __declspec(dllexport) -#else -WP_DECLARE_DATA -#endif - typedef enum { WP_RINGING = (1 << 0) } wp_flag_t; @@ -874,7 +849,7 @@ } -EX_DECLARE_DATA zap_module_t zap_module = { +zap_module_t zap_module = { "wanpipe", wanpipe_init, wanpipe_destroy, Michael Jerris wrote: > try the 3.5.2 drivers. > > Mike > > From mike at jerris.com Fri May 15 14:25:22 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 15 May 2009 17:25:22 -0400 Subject: [Freeswitch-users] FXS ports not working on Sangoma A200 [solved] In-Reply-To: <4A0DCB30.2030803@jacosoft.com> References: <4A0D995A.7060505@jacosoft.com> <72EC19D5-0359-4A34-A1C7-15557BA41921@jerris.com> <4A0DCB30.2030803@jacosoft.com> Message-ID: <6BD0CA84-4C6F-44FE-9CF2-D889A2B0DD46@jerris.com> sorry about the typo in that commit, this is fixed in tree now. Mike On May 15, 2009, at 4:06 PM, Brian Wood wrote: > Ah! I was looking at the dates not the version numbers on their site. > The beta is listed as 2008-xx-yy and the stable 2009-xx-yy, so I > grabbed > what I thought was the newer one. > > Once I rebuilt everything, FXO and FXS are both working. > > FYI: The last commit to ozmod_wanpipe.c broke it for Linux, but I > expect > someone will notice this out soon enough. I had to back down to the > previous revision to compile: > > Index: ozmod_wanpipe.c > =================================================================== > --- ozmod_wanpipe.c (revision 733) > +++ ozmod_wanpipe.c (revision 732) > @@ -36,35 +36,10 @@ > #include > #endif > #include "openzap.h" > -#ifndef __WINDOWS__ > #include > #include ... > > -#endif > #include "libsangoma.h" > > -#if defined(__WINDOWS__) > -/*! Backward compatible defines - current code is all using the old > names*/ > -#define sangoma_open_tdmapi_span_chan sangoma_open_api_span_chan > -#define sangoma_open_tdmapi_span sangoma_open_api_span > -#define sangoma_open_tdmapi_ctrl sangoma_open_api_ctrl > -#define sangoma_tdm_get_fe_status sangoma_get_fe_status > -#define sangoma_socket_close sangoma_close > -#define sangoma_tdm_get_hw_coding sangoma_get_hw_coding > -#define sangoma_tdm_set_fe_status sangoma_set_fe_status > -#define sangoma_tdm_get_link_status sangoma_get_link_status > -#define sangoma_tdm_flush_bufs sangoma_flush_bufs > -#define sangoma_tdm_cmd_exec sangoma_cmd_exec > -#define sangoma_tdm_read_event sangoma_read_event > -#define sangoma_readmsg_tdm sangoma_readmsg > -#define sangoma_readmsg_socket sangoma_readmsg > -#define sangoma_sendmsg_socket sangoma_writemsg > -#define sangoma_writemsg_tdm sangoma_writemsg > -#define sangoma_create_socket_intr sangoma_open_api_span_chan > -#define EX_DECLARE_DATA > __declspec(dllexport) > -#else > -WP_DECLARE_DATA > -#endif > - > typedef enum { > WP_RINGING = (1 << 0) > } wp_flag_t; > @@ -874,7 +849,7 @@ > } > > > -EX_DECLARE_DATA zap_module_t zap_module = { > +zap_module_t zap_module = { > "wanpipe", > wanpipe_init, > wanpipe_destroy, > > Michael Jerris wrote: >> try the 3.5.2 drivers. >> >> Mike >> >> > > > _ From living1108 at gmail.com Fri May 15 14:47:17 2009 From: living1108 at gmail.com (Errol Livingston) Date: Fri, 15 May 2009 17:47:17 -0400 Subject: [Freeswitch-users] Cisco 7905G IP Phone and Freeswitch Message-ID: <771c87020905151447td7e475albe7b00b480e552d4@mail.gmail.com> I've compiled/configured freeSwitch on Debian Etch - IP 192.168.24.10 on a computer at home behind a NATing firewall. My firewall/router has SIP (5080) and RTP (16834-) ports forwarded to the computer running freeSwitch. My setup is as follows: a) 2 Cisco 7905G IP phones configured with the latest firmware at extensions 1000 & 1004 b) X-Lite softphone at extension 1001. c) X-Lite softphone at extension 2010 at my office. Calls between ALL devices on my LAN (extensions 100x ) work without problems. I can receive calls from extension 2010 to the X_Lite extension (1001) on my LAN. My problem is that audio from external extension 2010 to either Cisco IP phones on my LAN is one-way. Cisco 7905G to X-Lite works but X-Lite to Cisco 7905G does not! The X-Lite phone at extension 2010 is also behind a NATing firewall! Has anyone experienced this issue before? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/a208ba1a/attachment.html From brian at freeswitch.org Fri May 15 15:19:50 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 15 May 2009 17:19:50 -0500 Subject: [Freeswitch-users] Cisco 7905G IP Phone and Freeswitch In-Reply-To: <771c87020905151447td7e475albe7b00b480e552d4@mail.gmail.com> References: <771c87020905151447td7e475albe7b00b480e552d4@mail.gmail.com> Message-ID: <0D04867F-78FE-4A23-A3C9-1A63D840F0B4@freeswitch.org> What port do these cisco's register with? /b On May 15, 2009, at 4:47 PM, Errol Livingston wrote: > a) 2 Cisco 7905G IP phones configured with the latest firmware at > extensions 1000 & 1004 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/89d16e85/attachment.html From jason at jasonjgw.net Fri May 15 17:53:05 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 16 May 2009 10:53:05 +1000 Subject: [Freeswitch-users] TLS initialization failures In-Reply-To: <20090429042219.GA5801@jdc.jasonjgw.net> References: <20090429042219.GA5801@jdc.jasonjgw.net> Message-ID: <20090516005305.GA8855@jdc.jasonjgw.net> I now have another core file from FreeSWITCH, generated when it was initializing the TLS during startup. I know this should all be in a bug report - I tried Jira again yesterday, but haven't been able to sort out the accessibility problems I was having with it. If you would be interested in a backtrace from this core file, let me know and I'll provide it as an e-mail attachment or use pastebin. The revision this time is 13266. From living1108 at gmail.com Fri May 15 18:40:37 2009 From: living1108 at gmail.com (Errol Livingston) Date: Fri, 15 May 2009 21:40:37 -0400 Subject: [Freeswitch-users] Cisco 7905G IP Phone and Freeswitch Message-ID: <771c87020905151840k5dfdb2b3q8637d215aee9a63d@mail.gmail.com> SIPPort : 5060 MediaPort: 16384 I've included ALL the 'SIP Parameters' below UID : 1004 PWD : **** Proxy : 192.168.24.10 AltProxyTimeOut : 0 UseLoginID : 0 LoginID : 1004 SIPRegInterval : 3600 MaxRedirect : 5 SIPRegOn : 1 NATIP : 0.0.0.0 SIPPort : 5060 MediaPort : 16384 OutBoundProxy : 0 MsgRetryLimits : 0x00000000 NatServer : 0 NatTimer : 0x00000000 DialPlan : 911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.- IPDialPlan : 1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090515/2745d751/attachment-0001.html From brian at freeswitch.org Fri May 15 19:54:33 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 15 May 2009 21:54:33 -0500 Subject: [Freeswitch-users] TLS initialization failures In-Reply-To: <20090516005305.GA8855@jdc.jasonjgw.net> References: <20090429042219.GA5801@jdc.jasonjgw.net> <20090516005305.GA8855@jdc.jasonjgw.net> Message-ID: <0DD8DD95-2FCE-465E-A78A-22367494FA46@freeswitch.org> Can you collect a backtrace for me... I have been testing TLS extensivley for the past two days and haven't seen a single crash yet!!! Just email it to me off list and I'll tlae care of it. Thanks Sent from my iPhone On May 15, 2009, at 7:53 PM, Jason White wrote: > I now have another core file from FreeSWITCH, generated when it was > initializing the TLS during startup. > > I know this should all be in a bug report - I tried Jira again > yesterday, but > haven't been able to sort out the accessibility problems I was > having with it. > > If you would be interested in a backtrace from this core file, let > me know and > I'll provide it as an e-mail attachment or use pastebin. > > The revision this time is 13266. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From yudha2008 at gmail.com Fri May 15 22:31:02 2009 From: yudha2008 at gmail.com (Baskar) Date: Sat, 16 May 2009 11:01:02 +0530 Subject: [Freeswitch-users] JavaScript session conference In-Reply-To: <87f2f3b90905151022x6cd999eeuf23ac95c8d57f706@mail.gmail.com> References: <4F928B52-FB3E-4192-B076-3ACF214AD10B@ken-ton.com> <87f2f3b90905140955y6758fdb5w61470affd43496e9@mail.gmail.com> <87f2f3b90905151022x6cd999eeuf23ac95c8d57f706@mail.gmail.com> Message-ID: Hi, I have designed a CRM in that caller will login and he will wait for the call. when the campaign start it get no from database and dial the number and bridge to the caller. > *Step1: I get Mobile Number and Extension Number from Database and pass >> those value to JavaScript. >> * > > > Who or what needs to get the mobile number and extension from a database? > What happens that leads to this process? > it will dial the Mobile number. Then caller will wait for the call once the Mobile no call is established caller will get the call and bridge the call when answer with is softphone. > > > >> * >> Step2: JavaScript will dial the both Mobile and Extension Number. After >> some time agent want to transfer the call to conference room. >> * > > > This is the first time I've heard mention of an agent. Sorry if I missed it > earlier. Who is the agent and what does he/she actually do? > Agent is nothing but the caller. He/she login and wait for the call > > > >> * >> Step3: Then agent will dial another Mobile Number and transfer the call >> into same conference room. >> * > > > What kind of telephone device will the agent be using? Hard phone, > softphone, etc.? > caller use softphone X-lite and SJPhone. > > > >> * >> I want the call Flow to work in this sequence >> >> 9841799874==>1001* > > > Who is extension 1001 in this scenario? > > 1001 is the caller who login and wait for the call > * >> >> Transfer both the call in to conference Room 3001 == 9841799874 + 1002 >> * > > > Who is extension 1002 in this scenario? > > sorry it is not 1002 it is 1001 i have give wrong. > * >> Agent call another Mobile Number >> >> 9841799852 >> >> Transfer the call in to same Conference Room 3001=== >> 9841799852 + 9841799874 + 1002 all the 3 Number should be in >> conference 3001.* >> >> *Can you assist me to do this above process through JavaScript >> >> Thanks in advance >> ** >> -- >> Warm Regards, >> N.Baskar >> * >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090516/6ff5b971/attachment.html From Prometheus001 at gmx.net Sat May 16 06:14:43 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 16 May 2009 15:14:43 +0200 Subject: [Freeswitch-users] Call For Participants: Lightning Talks at ClueCon 2009 In-Reply-To: <87f2f3b90905141314w3a2b24ccu41c692588555264b@mail.gmail.com> References: <87f2f3b90905141314w3a2b24ccu41c692588555264b@mail.gmail.com> Message-ID: <4A0EBC43.9060303@gmx.net> Hello Michael, I see that there are still some time slots available on 6th of Aug. I am thinking of doing a presentation on an application server and Web GUI for Fresswitch we have developed. Is it still possible to register for a full speaker slot? Best regards Peter Michael Collins schrieb: > *ClueCon 2009 is coming soon!* > > We are interested in your thoughts on subjects for lighting talks. We > would love to have a number of 5-10 minute presentations by members of > the community. If you would like to give a talk, or just have an idea > for a talk, please let us know. > > How do lightning talks work? Quite simply, the presenter has just a > few minutes to speak on a particular subject, usually no more than 10 > minutes. He or she will deliver the information rapidly, which means > keeping the presentation focused tightly on the subject being > discussed. Lightning talks usually do not have enough time for > audience Q&A. However, ClueCon has a long lunch period that is > designed to allow attendees plenty of time to interact. Those are > perfect times to discuss lightning talks or any other presentations. > Those who give presentations enjoy interacting with other attendees in > a relaxed atmosphere during lunch or in the evening at dinner. > > If you haven't already registered for ClueCon 2009 then please call us > at 877.742.CLUE right away and we will complete your registration. > Also, don't forget that expedia.com has some nice > hotel deals for the Wyndham Chicago. Book your room today! > > We look forward to hearing from you and seeing you all at ClueCon in > Chicago. > -Michael > http://www.cluecon.com > 877.742.CLUE > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dftoro at yahoo.com Sat May 16 10:11:57 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sat, 16 May 2009 10:11:57 -0700 (PDT) Subject: [Freeswitch-users] FS loses registration (Double Re-Register problem) Message-ID: <667347.32986.qm@web33504.mail.mud.yahoo.com> Greetings, ? I see Wireshark trace a double re-REGISTER in minus of? the 1 second between one and other REGISTER, then siemens pbx return "Status: 100 Trying??? (0 bindings)" and then FS loses registration. ? Wireshark trace:? http://pastebin.freeswitch.org/8978? ? 170 11:40:17.98 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 171 11:40:18.00 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 172 11:40:18.02 192.168.0.120 192.168.0.125 SIP Status: 200 OK??? (1 bindings) 365 11:42:15.69 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 366 11:42:15.69 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 367 11:42:15.72 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 368 11:42:15.72 192.168.0.120 192.168.0.125 SIP Status: 200 OK??? (1 bindings) 369 11:42:19.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 370 11:42:19.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 399 11:42:23.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 400 11:42:23.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 406 11:42:27.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 407 11:42:27.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 408 11:42:31.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 409 11:42:31.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 415 11:42:35.28 192.168.0.125 192.168.0.120 SIP Request: REGISTER sip:192.168.0.120;transport=udp 416 11:42:35.29 192.168.0.120 192.168.0.125 SIP Status: 100 Trying??? (0 bindings) 417 11:42:35.35 192.168.0.120 192.168.0.125 SIP Status: 408 Request Timeout??? (0 bindings) ? ?FreeSWITCH Version 1.0.4pre6 (13176) FS IP:???????????????? 192.168.0.125 siemes 3500 IP:? 192.168.0.120 ? I would appreciate any suggestions. ? Thank you ? Diego /09, Diego Toro wrote: From: Diego Toro Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users at lists.freeswitch.org Date: Friday, May 15, 2009, 7:58 AM I have been changing gateway parameters but i don't know what is the problem. ? any suggestion ? ? Diego --- On Thu, 5/14/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] FS loses registration To: freeswitch-users at lists.freeswitch.org Date: Thursday, May 14, 2009, 6:49 PM Seem to have left out a few things... /b On May 14, 2009, at 6:39 PM, Diego Toro wrote: I have a trouble, when FS start it register whit Siemens pbx, but after FS?loses registration.? I have a x-lite registered whit Siemens pbx and x-lite keeps register. ? I add whireshark trace? thank you ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090516/842e7509/attachment-0001.html From Prometheus001 at gmx.net Sat May 16 10:47:18 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 16 May 2009 19:47:18 +0200 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <87f2f3b90903051232l22db660blba57c46208f42d63@mail.gmail.com> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> <49B03402.8050601@gmx.net> <87f2f3b90903051232l22db660blba57c46208f42d63@mail.gmail.com> Message-ID: <4A0EFC26.6030706@gmx.net> Hello Michael, now some time later I did another try with the latest trunk. The problem were the grammar files fr the pizza demo. The old ones didn't work anymore with ne tnew version of pocketsphinx. Now with the new grammar files it works. I have updated the wiki accordingly. Best regards Peter Michael Collins schrieb: > On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX wrote: > >> Hello Brian, >> >> concerning >> >>> Well you should use ESL then ;) >>> >> I simply do not understand what you mean by this. Is it sarcastic? Am I >> asking stupid questions? >> >> > > ESL = Event Socket Library. It is an abstraction layer to make > interacting with the FS event socket a little easier. Look in the > source directory under libs/esl and you'll see all sorts of stuff. > Also check out the new-but-growing ESL wiki page: > > http://wiki.freeswitch.org/wiki/Esl > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Sat May 16 12:36:33 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 16 May 2009 12:36:33 -0700 Subject: [Freeswitch-users] Call For Participants: Lightning Talks at ClueCon 2009 In-Reply-To: <4A0EBC43.9060303@gmx.net> References: <87f2f3b90905141314w3a2b24ccu41c692588555264b@mail.gmail.com> <4A0EBC43.9060303@gmx.net> Message-ID: We are pretty much booked solid as we've got some unconfirmed speakers we haven't posted yet. I'm redoing the schedule and will have an updated one out this next week. One thing that we really need is backup speakers. Our experience is that there are always people who have emergencies and can't make it. Would you be willing to be one of our backups? There is a pretty good chance that you would speak but we won't know exactly which day or time. Please let me know what you think. -MC Sent from my iPhone On May 16, 2009, at 6:14 AM, Peter P GMX wrote: > Hello Michael, > > I see that there are still some time slots available on 6th of Aug. > I am > thinking of doing a presentation on an application server and Web GUI > for Fresswitch we have developed. > Is it still possible to register for a full speaker slot? > > Best regards > Peter > > > > Michael Collins schrieb: >> *ClueCon 2009 is coming soon!* >> >> We are interested in your thoughts on subjects for lighting talks. We >> would love to have a number of 5-10 minute presentations by members >> of >> the community. If you would like to give a talk, or just have an idea >> for a talk, please let us know. >> >> How do lightning talks work? Quite simply, the presenter has just a >> few minutes to speak on a particular subject, usually no more than 10 >> minutes. He or she will deliver the information rapidly, which means >> keeping the presentation focused tightly on the subject being >> discussed. Lightning talks usually do not have enough time for >> audience Q&A. However, ClueCon has a long lunch period that is >> designed to allow attendees plenty of time to interact. Those are >> perfect times to discuss lightning talks or any other presentations. >> Those who give presentations enjoy interacting with other attendees >> in >> a relaxed atmosphere during lunch or in the evening at dinner. >> >> If you haven't already registered for ClueCon 2009 then please call >> us >> at 877.742.CLUE right away and we will complete your registration. >> Also, don't forget that expedia.com has some >> nice >> hotel deals for the Wyndham Chicago. Book your room today! >> >> We look forward to hearing from you and seeing you all at ClueCon in >> Chicago. >> -Michael >> http://www.cluecon.com >> 877.742.CLUE >> --- >> --------------------------------------------------------------------- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nameer.kazzaz at gmail.com Sat May 16 15:53:03 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Sat, 16 May 2009 23:53:03 +0100 Subject: [Freeswitch-users] Dial-string to a registerd gatway Message-ID: <6c87b60905161553u44f72d6dj84dc30bc753d57a5@mail.gmail.com> Hi there, I'm using a quintum gateway for 16 pots phones, the gateway registers with freeswitch, calls into freeswitch work but, how do i dial into the gateway from freeswitch dialplan. My dialplan Can I get some help please ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090516/4fd9c48a/attachment.html From q.edward at gmail.com Sat May 16 16:12:19 2009 From: q.edward at gmail.com (Edward Q.) Date: Sat, 16 May 2009 19:12:19 -0400 Subject: [Freeswitch-users] Dial-string to a registerd gatway In-Reply-To: <6c87b60905161553u44f72d6dj84dc30bc753d57a5@mail.gmail.com> References: <6c87b60905161553u44f72d6dj84dc30bc753d57a5@mail.gmail.com> Message-ID: <89313a90905161612u6b7b76carb3cf360e1b3353aa@mail.gmail.com> Create your external dialplan /conf/dialplan/default/xx_xxxx.xml and then create /conf/sip_profiles/external/xxxx.xml your gateway xml. Thenk you can use your gaetway. Ed On Sat, May 16, 2009 at 6:53 PM, Nameer Kazzaz wrote: > Hi there, > I'm using a quintum gateway for 16 pots phones, the gateway registers with > freeswitch, calls into freeswitch work but, how do i dial into the gateway > from freeswitch dialplan. My dialplan > > > > Can I get some help please ?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090516/639f2fc5/attachment.html From adam.falcone at gmail.com Sat May 16 19:10:14 2009 From: adam.falcone at gmail.com (adamF) Date: Sat, 16 May 2009 19:10:14 -0700 (PDT) Subject: [Freeswitch-users] Running FreeSwitch in the background In-Reply-To: References: <8ECA0A5A-E37A-4B45-ADE7-81060CC63119@gmail.com> Message-ID: <23579550.post@talk.nabble.com> Yes I am passing -nc when starting freeswitch and I can receive calls without issue initially. If I wait 10-15 then try to place another incoming call freeswitch will not pick it up. I haaven't found where anyone else has reported this issue so I am at a loss. Ken Rice-2 wrote: > > How are you running it in the background? > > Freeswitch -nc starts it in the background and it should run fine there > > > >> From: Adam Falcone >> Reply-To: >> Date: Sun, 10 May 2009 10:20:32 -0700 >> To: >> Subject: [Freeswitch-users] Running FreeSwitch in the background >> >> after starting freeswitch I am able to receive incoming calls but if I >> let freeswitch run in the background it stops picking up incoming >> calls. The only way I am able to get freeswitch to recognize incoming >> calls again is by making an outbound call and it then afterwards >> receives inbound calls. Not sure what if any log information would be >> beneficial to post here. I am new to freeswitch. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Running-FreeSwitch-in-the-background-tp23472194p23579550.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Sat May 16 20:14:15 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 17 May 2009 13:14:15 +1000 Subject: [Freeswitch-users] Running FreeSwitch in the background In-Reply-To: <23579550.post@talk.nabble.com> References: <8ECA0A5A-E37A-4B45-ADE7-81060CC63119@gmail.com> <23579550.post@talk.nabble.com> Message-ID: <20090517031415.GA25176@jdc.jasonjgw.net> adamF wrote: > > Yes I am passing -nc when starting freeswitch and I can receive calls without > issue initially. If I wait 10-15 then try to place another incoming call > freeswitch will not pick it up. I haaven't found where anyone else has > reported this issue so I am at a loss. Is there a NAT device between your FreeSWITCH system and the SIP endpoints that are making the calls? I'm assuming these are SIP calls; if not, I would suggest that you explain the scenario in greater detail. If there is a NAT device involved, it could well be your problem, and you could try setting a ping parameter in the relevant FreeSWITCH profile to keep the NAT port mapping alive. This might well be a problem that you'll have to solve by reconfiguring the NAT device (a router, for example), if there is one that is responsible for your issues. From nameer.kazzaz at gmail.com Sun May 17 00:35:45 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Sun, 17 May 2009 08:35:45 +0100 Subject: [Freeswitch-users] Dial-string to a registerd gatway In-Reply-To: <89313a90905161612u6b7b76carb3cf360e1b3353aa@mail.gmail.com> References: <6c87b60905161553u44f72d6dj84dc30bc753d57a5@mail.gmail.com> <89313a90905161612u6b7b76carb3cf360e1b3353aa@mail.gmail.com> Message-ID: <4A0FBE51.7050807@gmail.com> Hi there, No that is not my problem. I don't need an external profile I only have one sip profile and a directory of useres. So my gateway is a user.xml because it registers to freeswitch. For this setup I don't need external.xml. Thanks Edward Q. wrote: > Create your external dialplan /conf/dialplan/default/xx_xxxx.xml and > then create /conf/sip_profiles/external/xxxx.xml your gateway xml. > Thenk you can use your gaetway. > Ed > > > > > On Sat, May 16, 2009 at 6:53 PM, Nameer Kazzaz > > wrote: > > Hi there, > I'm using a quintum gateway for 16 pots phones, the gateway > registers with freeswitch, calls into freeswitch work but, how do > i dial into the gateway from freeswitch dialplan. My dialplan > > data="user/${dialed_extension}@${domain_name}"/> > > Can I get some help please ?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Sun May 17 10:00:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 17 May 2009 19:00:45 +0200 Subject: [Freeswitch-users] Call For Participants: Lightning Talks at ClueCon 2009 In-Reply-To: References: <87f2f3b90905141314w3a2b24ccu41c692588555264b@mail.gmail.com> <4A0EBC43.9060303@gmx.net> Message-ID: <4A1042BD.5020101@gmx.net> Hello Michael, I will have to fly in from Germany. So if there's 90% chance to speak, fine. If there's only 50% chance, ??? How would you rate the chance? Best regards Peter Michael S Collins schrieb: > We are pretty much booked solid as we've got some unconfirmed speakers > we haven't posted yet. I'm redoing the schedule and will have an > updated one out this next week. One thing that we really need is > backup speakers. Our experience is that there are always people who > have emergencies and can't make it. Would you be willing to be one of > our backups? There is a pretty good chance that you would speak but we > won't know exactly which day or time. > > Please let me know what you think. > -MC > > Sent from my iPhone > > On May 16, 2009, at 6:14 AM, Peter P GMX wrote: > > >> Hello Michael, >> >> I see that there are still some time slots available on 6th of Aug. >> I am >> thinking of doing a presentation on an application server and Web GUI >> for Fresswitch we have developed. >> Is it still possible to register for a full speaker slot? >> >> Best regards >> Peter >> >> >> >> Michael Collins schrieb: >> >>> *ClueCon 2009 is coming soon!* >>> >>> We are interested in your thoughts on subjects for lighting talks. We >>> would love to have a number of 5-10 minute presentations by members >>> of >>> the community. If you would like to give a talk, or just have an idea >>> for a talk, please let us know. >>> >>> How do lightning talks work? Quite simply, the presenter has just a >>> few minutes to speak on a particular subject, usually no more than 10 >>> minutes. He or she will deliver the information rapidly, which means >>> keeping the presentation focused tightly on the subject being >>> discussed. Lightning talks usually do not have enough time for >>> audience Q&A. However, ClueCon has a long lunch period that is >>> designed to allow attendees plenty of time to interact. Those are >>> perfect times to discuss lightning talks or any other presentations. >>> Those who give presentations enjoy interacting with other attendees >>> in >>> a relaxed atmosphere during lunch or in the evening at dinner. >>> >>> If you haven't already registered for ClueCon 2009 then please call >>> us >>> at 877.742.CLUE right away and we will complete your registration. >>> Also, don't forget that expedia.com has some >>> nice >>> hotel deals for the Wyndham Chicago. Book your room today! >>> >>> We look forward to hearing from you and seeing you all at ClueCon in >>> Chicago. >>> -Michael >>> http://www.cluecon.com >>> 877.742.CLUE >>> --- >>> --------------------------------------------------------------------- >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From adam.falcone at gmail.com Sun May 17 10:38:37 2009 From: adam.falcone at gmail.com (adamF) Date: Sun, 17 May 2009 10:38:37 -0700 (PDT) Subject: [Freeswitch-users] Running FreeSwitch in the background In-Reply-To: <20090517031415.GA25176@jdc.jasonjgw.net> References: <8ECA0A5A-E37A-4B45-ADE7-81060CC63119@gmail.com> <23579550.post@talk.nabble.com> <20090517031415.GA25176@jdc.jasonjgw.net> Message-ID: <23585691.post@talk.nabble.com> I was able to fix this by assigning my public IP to the following in vars.xml XX's obviously being my public IP and now Freeswitch is receiving calls while running in the background. Thanks for your help Jason White-14 wrote: > > adamF wrote: >> >> Yes I am passing -nc when starting freeswitch and I can receive calls >> without >> issue initially. If I wait 10-15 then try to place another incoming call >> freeswitch will not pick it up. I haaven't found where anyone else has >> reported this issue so I am at a loss. > > Is there a NAT device between your FreeSWITCH system and the SIP endpoints > that are making the calls? I'm assuming these are SIP calls; if not, I > would > suggest that you explain the scenario in greater detail. > > If there is a NAT device involved, it could well be your problem, and you > could try setting a ping parameter in the relevant FreeSWITCH profile to > keep > the NAT port mapping alive. > > This might well be a problem that you'll have to solve by reconfiguring > the > NAT device (a router, for example), if there is one that is responsible > for > your issues. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Running-FreeSwitch-in-the-background-tp23472194p23585691.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From adam.falcone at gmail.com Sun May 17 11:12:20 2009 From: adam.falcone at gmail.com (adamF) Date: Sun, 17 May 2009 11:12:20 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Drops Call after 30 Seconds Message-ID: <23586016.post@talk.nabble.com> I am having an issue where freeswitch drops an incoming call after being connected for 30 seconds. Here is my console output starting from the termination sequence. Any help would be greatly appreciated. 2009-05-17 10:59:41 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/2062842721 at 64.24.35.74 entering state [terminating] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:355 audio_bridge_thread() sofia/external/2062842721 at 64.24.35.74 ending bridge by request from write function 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/external/2062842721 at 64.24.35.74 [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:471 audio_bridge_on_exchange_media() Hangup sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:467 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) State EXCHANGE_MEDIA going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 4 (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) Locked, Waiting on external entities 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/external/2062842721 at 64.24.35.74 receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/external/2062842721 at 64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/external/2062842721 at 64.24.35.74] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:960 switch_ivr_multi_threaded_bridge() Hangup sofia/external/2062842721 at 64.24.35.74 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/2062842721 at 64.24.35.74 [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/external/2062842721 at 64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) State EXECUTE going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/2062842721 at 64.24.35.74 hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/2062842721 at 64.24.35.74 Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 3 (sofia/external/2062842721 at 64.24.35.74) Locked, Waiting on external entities 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 3 (sofia/external/2062842721 at 64.24.35.74) Ended 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/2062842721 at 64.24.35.74 [CS_HANGUP] 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 4 (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes) Ended 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat=yes [CS_HANGUP] -- View this message in context: http://www.nabble.com/Freeswitch-Drops-Call-after-30-Seconds-tp23586016p23586016.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ronmccar at gmail.com Sun May 17 11:26:05 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Sun, 17 May 2009 11:26:05 -0700 Subject: [Freeswitch-users] Logging 503's or other errors Message-ID: <3885f4fe0905171126y7eb9b419x4c103a53cf4a6cad@mail.gmail.com> Hi list, Ive been trying to find a way to log 503's, 480's and other SIP response codes. If we have continue_on_fail=true and have multiple gateways for the call to go out, if the 1st,2nd or whatever gateways fail can we log it somehow? We'd like to know if a carrier is having issues or not letting us send calls for some reason, from what I can tell I only show one CDR get written and that's at the end of the call, so it says nothing about the gateways we tried to send a call before and if they failed. Any ideals on how to do this? Im using the XML CURL dialplan if that matter. Any ideals how this could be setup so we can keep track of what is going on? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090517/3902f6a3/attachment.html From ronmccar at gmail.com Sun May 17 11:33:26 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Sun, 17 May 2009 11:33:26 -0700 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <49A39020.3020808@post.cz> References: <499E92E4.5010503@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> <49A39020.3020808@post.cz> Message-ID: <3885f4fe0905171133qc993016ibda9555a71f75661@mail.gmail.com> Kokoska, Did you ever find a solution for this? I have been working on this as well, trying to write some perl application to read the data from ngrep and parse it, but have got no where. I hope you have better luck then I have! On Mon, Feb 23, 2009 at 11:13 PM, kokoska.rokoska wrote: > Joseph Bajin napsal(a): > > If you write it correctly it will work just fine. > > Yes, this is challenge I have talked about :-) > > > That is how most of > > all the other correlation engines work. > > I don't have enough informations but from what I heard from friendly > "competitors" they are usualy log (SIP|ISUP) messages after they are > parsed by their "routing" servers and not run separate > tshark+parser+logger. Or they duplicate (just) SIP messages to separate > machine and parse and log them there (SERlike server + sip_trace). > > > Your setup is not going to be > > bigger than some of the large telecoms that use these systems today. > > > > I hope so :-) > > > Thanks once more, Joseph, for your info! > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090517/a73fb35e/attachment.html From kokoska.rokoska at post.cz Sun May 17 14:41:32 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Sun, 17 May 2009 23:41:32 +0200 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <3885f4fe0905171133qc993016ibda9555a71f75661@mail.gmail.com> References: <499E92E4.5010503@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> <49A39020.3020808@post.cz> <3885f4fe0905171133qc993016ibda9555a71f75661@mail.gmail.com> Message-ID: <4A10848C.6000607@post.cz> Ron McCarthy napsal(a): > Kokoska, > > Did you ever find a solution for this? I have been working on this as > well, trying to write some perl application to read the data from ngrep > and parse it, but have got no where. > > I hope you have better luck then I have! > I'm sorry, Ron, but I still use "workaround" (Kamailio + sip_trace) to log SIP messages to DB: 1. It works :-) Even with about 10k+ INSERTs/UPDATEs per second. 2. It is simple to setup. 3. I'm not good programmer to write working and fast enough "parser" :-) BTW: I don't know what your setup/load is, but I doubt if perl can work for you. Just for curiosity I tried to run sipgrep (ngrep + perl parser) and it "missed" 90 % of SIP messages and "eats" all HW resources... Best regards, kokoska.rokoska From helmut.kuper at ewetel.de Mon May 18 00:33:59 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 18 May 2009 09:33:59 +0200 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <3885f4fe0905171133qc993016ibda9555a71f75661@mail.gmail.com> References: <499E92E4.5010503@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> <49A39020.3020808@post.cz> <3885f4fe0905171133qc993016ibda9555a71f75661@mail.gmail.com> Message-ID: <4A110F67.9020707@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, maybe this helps: http://www.wesip.com/mediawiki/index.php/SipSpy regards helmut On 17.05.2009 20:33, Ron McCarthy wrote: > Kokoska, > > Did you ever find a solution for this? I have been working on this as > well, trying to write some perl application to read the data from ngrep > and parse it, but have got no where. > > I hope you have better luck then I have! > > > > > > On Mon, Feb 23, 2009 at 11:13 PM, kokoska.rokoska > > wrote: > > Joseph Bajin napsal(a): > > If you write it correctly it will work just fine. > > Yes, this is challenge I have talked about :-) > > > That is how most of > > all the other correlation engines work. > > I don't have enough informations but from what I heard from friendly > "competitors" they are usualy log (SIP|ISUP) messages after they are > parsed by their "routing" servers and not run separate > tshark+parser+logger. Or they duplicate (just) SIP messages to separate > machine and parse and log them there (SERlike server + sip_trace). > > > Your setup is not going to be > > bigger than some of the large telecoms that use these systems today. > > > > I hope so :-) > > > Thanks once more, Joseph, for your info! > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKEQ9m4tZeNddg3dwRAofUAJ4nRE8237OQQyb2ybzDiAyH4XEYIgCffmry QDyWbwRq/0IHQTRW+i/yhH4= =w3UC -----END PGP SIGNATURE----- From pawzlion at gmail.com Mon May 18 01:20:46 2009 From: pawzlion at gmail.com (David Robinson) Date: Mon, 18 May 2009 18:20:46 +1000 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user Message-ID: Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ machine on my modem so it can receive incoming connections without any NAT related problems. I'm trying to get a user outside on the internet to connect to my FS box and register as an internal user. He is using X-Lite on his laptop behind his own NAT. His external IP is 203.206.171.118. His registration looks like this: Call-ID: NGU1NjE4ZmIxODMwMWM1OTZlMDYzMzk0NjMyMjM1YWY. User: 1001 at 10.0.0.12 Contact: "124.254.81.250" Agent: X-Lite release 1014k stamp 47051 Status: Registered(UDP)(unknown) EXP(2009-05-18 19:32:03) Host: kira IP: 203.206.171.118 Port: 40168 Auth-User: 1001 Auth-Realm: 124.254.81.250 I note that it's registered as plain UDP, not UDP-NAT like my own internal extensions are. The dialplan is set to route this DID (0746029001) to user 1001@$$ {domain} as follows: When I try and make a call from my mobile (0451282630) to the DID, it says it's bridging to USER/1001 at 10.0.0.12, but when the person answers, we get no audio in either direction. It rings and answers fine, it just doesn't send any audio in either direction so I'm suspecting a bridging problem. The log file of the connection is on the web at http://pastebin.freeswitch.org/8990 The bridge line is: EXECUTE sofia/external/0451282630 at 203.161.130.132 bridge(USER/1001 at 10.0.0.12 ) But the sofia address for the connection is shown as sofia/internal/sip:1001 at 203.206.171.118:40168;rinstance=c5779e159bbe8bc7 Is this correct ? Am I missing something fundamental ? His user address is @10.0.0.12, but his sofia address is sip: 1001 at 203.206.171.118. Is this OK or should his user ID be at his actual ip address ? This seems normal to me as I believe the 10.0.0.12 address is the "domain" of the FS box. Is it OK that he's in the same domain as my own users on my LAN or am I supposed to configure a different domain for him because he's "outside". I thought maybe it was a double-NAT problem, but the log doesn't show any fs_nat=yes entries so I assume it's not trying to NAT him (as it shouldn't). The situation is an external mobile rings my DID, so the call comes in from my provider's address, hits my FS box, which successfully sends at least the ringing information out to his softphone at his external IP, but then when it bridges, it seems not to send the audio to the right place. I'm terrible with FS log files so I have no idea whether any of the entries are wrong. What's likely to be my issue here ? Is it NAT- related, or routing related ? Any suggestions appreciated. David From fdelawarde at wirelessmundi.com Mon May 18 01:29:30 2009 From: fdelawarde at wirelessmundi.com (Francois Delawarde) Date: Mon, 18 May 2009 10:29:30 +0200 Subject: [Freeswitch-users] openzap and progress detection Message-ID: <1242635370.17063.67.camel@localhost.localdomain> Hello, I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO +1xFXS), and am trying first to make the FXO work with Openzap and Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and loads the spans, but I'm currently enable to dial out with the FXO module, it doesn't dial anything and times-out after 30 seconds. I believe it has to do with some tone detection and therefore have a few questions: - When I plug in the line, dahdi sends an event (event 17) that is ignored by OZ. Can we enable some type of battery check in OZ before dialing out, or is there some variable to monitor battery (oz dump doesn't show the battery status)? - Does OZ use the polarity switch events sent by dahdi (in kewlstart mode) for answer and hangup detection? - Apparently, OZ does tone progress detection by frequency, but here in Spain, most tones use the same 425Hz frequency with different on-off timing. Is it possible to detect those? - As some PBX in Spain transfer calls by first hanging up and picking up on another phone, can we enable/disable parts of polarity switch and/or tone progress detection (ex: (hangup)/(answer)onpolarityswitch in Asterisk) - Some lines here are connected to very old FXS from operators that have low sound quality and can take a few seconds to give a dial tone when picking up. Is it possible to introduce a delay before sending DTMF digits when dialing? Is it possible to "relax" DTMF detection, and tweak DTMF settings (make them a bit longer, with a longer pause for the other side to detect)? My "ideal" case to make it work in every case around here would be to: - have OZ fail to dial if battery is not present (and be able to fetch battery status somehow) - disable tone progress (sometimes call ends up on some local PBX that answers and provides US tones which are different) - be able to have an initial pause before dialing with DTMF digits - use polarity switch to detect remote answer, but not hangup (for transfer issues) Is the above possible? Thanks in advance, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/23cd173e/attachment.html From jason at jasonjgw.net Mon May 18 01:45:43 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 18 May 2009 18:45:43 +1000 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user In-Reply-To: References: Message-ID: <20090518084543.GA32537@jdc.jasonjgw.net> David Robinson wrote: > Is this correct ? Am I missing something fundamental ? My suspicion is that the RTP traffic isn't traversing the NAT properly. You may have to configure the routers at both ends to forward the RTP packets to the correct destinations. There is a good discussion of NAT on the wiki. One of the great advantages of IPv6 is that NAT goes away altogether. I've been achieving quite reasonable call quality even across IPv6-over-IPv4 tunnels. From andy at fabulous4.co.uk Mon May 18 02:07:37 2009 From: andy at fabulous4.co.uk (Andy) Date: Mon, 18 May 2009 10:07:37 +0100 Subject: [Freeswitch-users] Freeswitch Drops Call after 30 Seconds In-Reply-To: <23586016.post@talk.nabble.com> References: <23586016.post@talk.nabble.com> Message-ID: Hi Adam, We had exactly the same problem which we initially believed to be a NAT firewall issue. However, I changed my firewall to transparent mode and the problem still persisted. In the end, I solved the problem by changing VOIP provider. I was using AQL which I couldn't make work and now use VOIPTALK. Cheers Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of adamF Sent: 17 May 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch Drops Call after 30 Seconds I am having an issue where freeswitch drops an incoming call after being connected for 30 seconds. Here is my console output starting from the termination sequence. Any help would be greatly appreciated. 2009-05-17 10:59:41 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/2062842721 at 64.24.35.74 entering state [terminating] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:355 audio_bridge_thread() sofia/external/2062842721 at 64.24.35.74 ending bridge by request from write function 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/external/2062842721 at 64.24.35.74 [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:471 audio_bridge_on_exchange_media() Hangup sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:467 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) State EXCHANGE_MEDIA going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 4 (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) Locked, Waiting on external entities 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/external/2062842721 at 64.24.35.74 receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/external/2062842721 at 64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/external/2062842721 at 64.24.35.74] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:960 switch_ivr_multi_threaded_bridge() Hangup sofia/external/2062842721 at 64.24.35.74 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/2062842721 at 64.24.35.74 [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/external/2062842721 at 64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) State EXECUTE going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/2062842721 at 64.24.35.74 hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/2062842721 at 64.24.35.74 Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/2062842721 at 64.24.35.74) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 3 (sofia/external/2062842721 at 64.24.35.74) Locked, Waiting on external entities 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 3 (sofia/external/2062842721 at 64.24.35.74) Ended 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/2062842721 at 64.24.35.74 [CS_HANGUP] 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 4 (sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) Ended 2009-05-17 10:59:41 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/sip:1000 at 192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [CS_HANGUP] -- View this message in context: http://www.nabble.com/Freeswitch-Drops-Call-after-30-Seconds-tp23586016p2358 6016.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jim at evolutiontel.net Mon May 18 04:22:56 2009 From: jim at evolutiontel.net (Jim Burke) Date: Mon, 18 May 2009 21:22:56 +1000 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user In-Reply-To: References: Message-ID: Can you post the INVITE and 200 OK messages from your mates end of the call. Even if you forward the ports on the router, the RTP will not traverse correctly if the advertised IP address is an internal one for both ends. On Mon, May 18, 2009 at 6:20 PM, David Robinson wrote: > Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ > machine on my modem so it can receive incoming connections without any > NAT related problems. > > I'm trying to get a user outside on the internet to connect to my FS > box and register as an internal user. He is using X-Lite on his laptop > behind his own NAT. His external IP is 203.206.171.118. > > His registration looks like this: > > Call-ID: ? ? ? ?NGU1NjE4ZmIxODMwMWM1OTZlMDYzMzk0NjMyMjM1YWY. > User: ? ? ? ? ? 1001 at 10.0.0.12 > Contact: ? ? ? ?"124.254.81.250" ?> > Agent: ? ? ? ? ?X-Lite release 1014k stamp 47051 > Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-05-18 19:32:03) > Host: ? ? ? ? ? kira > IP: ? ? ? ? ? ? 203.206.171.118 > Port: ? ? ? ? ? 40168 > Auth-User: ? ? ?1001 > Auth-Realm: ? ? 124.254.81.250 > > I note that it's registered as plain UDP, not UDP-NAT like my own > internal extensions are. > > The dialplan is set to route this DID (0746029001) to user 1001@$$ > {domain} as follows: > > ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? ? ? ? ? > ? ? ? ? ? ? ? ? > ? ? ? ? > > When I try and make a call from my mobile (0451282630) to the DID, it > says it's bridging to USER/1001 at 10.0.0.12, but when the person > answers, we get no audio in either direction. It rings and answers > fine, it just doesn't send any audio in either direction so I'm > suspecting a bridging problem. > > The log file of the connection is on the web at http://pastebin.freeswitch.org/8990 > > The bridge line is: > EXECUTE sofia/external/0451282630 at 203.161.130.132 bridge(USER/1001 at 10.0.0.12 > ) > > But the sofia address for the connection is shown as sofia/internal/sip:1001 at 203.206.171.118:40168;rinstance=c5779e159bbe8bc7 > > Is this correct ? Am I missing something fundamental ? His user > address is @10.0.0.12, but his sofia address is sip: > 1001 at 203.206.171.118. Is this OK or should his user ID be at his > actual ip address ? This seems normal to me as I believe the 10.0.0.12 > address is the "domain" of the FS box. Is it OK that he's in the same > domain as my own users on my LAN or am I supposed to configure a > different domain for him because he's "outside". > > I thought maybe it was a double-NAT problem, but the log doesn't show > any fs_nat=yes entries so I assume it's not trying to NAT him (as it > shouldn't). The situation is an external mobile rings my DID, so the > call comes in from my provider's address, hits my FS box, which > successfully sends at least the ringing information out to his > softphone at his external IP, but then when it bridges, it seems not > to send the audio to the right place. > > I'm terrible with FS log files so I have no idea whether any of the > entries are wrong. What's likely to be my issue here ? Is it NAT- > related, or routing related ? Any suggestions appreciated. > > David > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From helmut.kuper at ewetel.de Mon May 18 05:41:44 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 18 May 2009 14:41:44 +0200 Subject: [Freeswitch-users] ODBC and Core-DB Message-ID: <4A115788.9080709@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, does anybody know if and how FS can export its core db to an external database via odbc like mod_limit or mod_sofia? If not, is such a feature planned for the near future? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKEVeI4tZeNddg3dwRAuMEAJ0dWJYGNDVDdtqJJuBRU/MFAtfAYQCeMqkR 7jEHwH2bxWEb/ccbdajxt5U= =XYMu -----END PGP SIGNATURE----- From mrene_lists at avgs.ca Mon May 18 05:46:50 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 18 May 2009 14:46:50 +0200 Subject: [Freeswitch-users] ODBC and Core-DB In-Reply-To: <4A115788.9080709@ewetel.de> References: <4A115788.9080709@ewetel.de> Message-ID: <10EA7A8F-4F3F-4FB3-9685-E1B2465D2BCD@avgs.ca> No, you can't. Math On 18-May-09, at 2:41 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > does anybody know if and how FS can export its core db to an external > database via odbc like mod_limit or mod_sofia? If not, is such a > feature > planned for the near future? > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKEVeI4tZeNddg3dwRAuMEAJ0dWJYGNDVDdtqJJuBRU/MFAtfAYQCeMqkR > 7jEHwH2bxWEb/ccbdajxt5U= > =XYMu > -----END PGP SIGNATURE----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon May 18 06:05:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 18 May 2009 08:05:07 -0500 Subject: [Freeswitch-users] Logging 503's or other errors In-Reply-To: <3885f4fe0905171126y7eb9b419x4c103a53cf4a6cad@mail.gmail.com> References: <3885f4fe0905171126y7eb9b419x4c103a53cf4a6cad@mail.gmail.com> Message-ID: <191c3a030905180605w31b6114aoceabb866108b2a7@mail.gmail.com> enable the b leg cdr as well and you will also get cdr from the b leg perspective. both xml cdr and cdr csv have params in the config to enable it. On Sun, May 17, 2009 at 1:26 PM, Ron McCarthy wrote: > Hi list, > > Ive been trying to find a way to log 503's, 480's and other SIP response > codes. If we have continue_on_fail=true and have multiple gateways for the > call to go out, if the 1st,2nd or whatever gateways fail can we log it > somehow? We'd like to know if a carrier is having issues or not letting us > send calls for some reason, from what I can tell I only show one CDR get > written and that's at the end of the call, so it says nothing about the > gateways we tried to send a call before and if they failed. > > Any ideals on how to do this? Im using the XML CURL dialplan if that > matter. Any ideals how this could be setup so we can keep track of what is > going on? > > Thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/de8f2be1/attachment-0001.html From 12ukwn at gmail.com Sat May 16 09:31:22 2009 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 16 May 2009 18:31:22 +0200 Subject: [Freeswitch-users] silly (?) questions Message-ID: <20090516183122.05495675@osiris.defcon1> Hi list, I just discovered FS (practiced a bit * 2 years ago, but too much unstable) and find it cool, NOT CPU greedy and (almost) working ouf of the web. I'd like to know if "star codes" (such as *98) are normalized or not? (and if so, where I could find a list) Also, as I don't use very much my phone and mostly don't pay for it (I live in france and got unlimited free call for 70-80 countries, and my phone is actually plugged in my ADSL box) I'd like to leave access for other people through something like DUNDi (that I don't really know.) BUT not everything is free (i.e.: cellular phones calls cost ?0.22 @ connection + ?0.22/min); thus I must forbid this kind of calls. Does anybody have realized that, because I need a good template? Thanks JY -- Old mail has arrived. From kiste at kiste.org Sun May 17 05:07:29 2009 From: kiste at kiste.org (Uwe Kastens) Date: Sun, 17 May 2009 14:07:29 +0200 Subject: [Freeswitch-users] freeswitch and invites without sdp in initial invite Message-ID: <4A0FFE01.4090900@kiste.org> Hi there, Freeswitch looks really interesting. I am trying to connect a softswitch wich does some strange things. If a call arrives from pots the softswitch won't send sdp information in the initial invite. Is this something I can change in the sip-profile with freeswitch? BR Uwe From kiste at kiste.org Sun May 17 10:45:08 2009 From: kiste at kiste.org (Uwe Kastens) Date: Sun, 17 May 2009 19:45:08 +0200 Subject: [Freeswitch-users] freeswitch and invites without sdp in initial invite Message-ID: <4A104D24.1010009@kiste.org> Hi there, Freeswitch looks really interesting. I am trying to connect a softswitch wich does some strange things. If a call arrives from pots the softswitch won't send sdp information in the initial invite. Is this something I can change in the sip-profile with freeswitch? BR Uwe From kiste at kiste.org Mon May 18 06:38:08 2009 From: kiste at kiste.org (Uwe Kastens) Date: Mon, 18 May 2009 15:38:08 +0200 Subject: [Freeswitch-users] fixed Re: freeswitch and invites without sdp in initial invite In-Reply-To: <4A104D24.1010009@kiste.org> References: <4A104D24.1010009@kiste.org> Message-ID: <4A1164C0.8030500@kiste.org> Hi, allowing 3cc fixes the problem. BR Uwe Uwe Kastens schrieb: > Hi there, > > Freeswitch looks really interesting. I am trying to connect a softswitch > wich does some strange things. If a call arrives from pots the > softswitch won't send sdp information in the initial invite. > > Is this something I can change in the sip-profile with freeswitch? > > BR > > Uwe > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- kiste lat: 54.322684, lon: 10.13586 From Prometheus001 at gmx.net Mon May 18 07:04:43 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 18 May 2009 16:04:43 +0200 Subject: [Freeswitch-users] silly (?) questions In-Reply-To: <20090516183122.05495675@osiris.defcon1> References: <20090516183122.05495675@osiris.defcon1> Message-ID: <4A116AFB.9050300@gmx.net> Hello Jean-Yves, did you ever try a call-trough? (a person dials in (1234567, see below) types the target number as DTMF and gets connected to this number? A basic dialplan can be like this: For "\d+" you may define your regular expression, which numbers you would accept. Also you may try to redirect into the dialplan again after the number is entered (instead of directly transferring the call). Best regards Peter Jean-Yves F. Barbier schrieb: > Hi list, > > I just discovered FS (practiced a bit * 2 years ago, but too much unstable) > and find it cool, NOT CPU greedy and (almost) working ouf of the web. > > I'd like to know if "star codes" (such as *98) are normalized or not? > (and if so, where I could find a list) > > Also, as I don't use very much my phone and mostly don't pay for it (I > live in france and got unlimited free call for 70-80 countries, and > my phone is actually plugged in my ADSL box) I'd like to leave access > for other people through something like DUNDi (that I don't really > know.) > BUT not everything is free (i.e.: cellular phones calls cost ?0.22 @ > connection + ?0.22/min); thus I must forbid this kind of calls. > > Does anybody have realized that, because I need a good template? > > Thanks > > JY > From andy at fabulous4.co.uk Mon May 18 09:21:54 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 18 May 2009 17:21:54 +0100 Subject: [Freeswitch-users] crash-protection and monit Message-ID: <148FF9B72A934BF1B793121735D1499C@D810> Hi, Is there any reason why the crash-protection parameter in switch.conf.xml defaults to false and are there any downsides to setting it to true? The documentation says it helps with certain types of crashes, can anyone tell me what sort of crashes in particular it helps to prevent as my freeswitch install seems to crash every few days. Also, does anyone have an example of the monit setup for freeswitch to restart it when it fails? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/b46bc931/attachment.html From mrene_lists at avgs.ca Mon May 18 09:27:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 18 May 2009 18:27:52 +0200 Subject: [Freeswitch-users] crash-protection and monit In-Reply-To: <148FF9B72A934BF1B793121735D1499C@D810> References: <148FF9B72A934BF1B793121735D1499C@D810> Message-ID: <3630465C-E475-49B1-A8DF-60BCC97D7634@avgs.ca> Hi, Crash protection catches segmentation faul signals and try to kill that thread only. It works for "stupid" errors like a null pointer dereference, but in most scenarios, a crash means something in the process memory was corrupted. Ignoring it will just make it crash later on, thats why the default is false. Now if you have a crash, you should update to svn trunk and if it still happens, report it on Jira ( see http://wiki.freeswitch.org/wiki/Reporting_Bugs ) so someone can look at it and fix it. Math On 18-May-09, at 6:21 PM, Andy Ayers wrote: > Hi, > > Is there any reason why the crash-protection parameter in > switch.conf.xml defaults to false and are there any downsides to > setting it to true? The documentation says it helps with certain > types of crashes, can anyone tell me what sort of crashes in > particular it helps to prevent as my freeswitch install seems to > crash every few days. > > Also, does anyone have an example of the monit setup for freeswitch > to restart it when it fails? > > Many thanks > Andy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/481bf0f6/attachment.html From freeswitch-users-list at metik.com Mon May 18 11:08:18 2009 From: freeswitch-users-list at metik.com (Metik) Date: Mon, 18 May 2009 14:08:18 -0400 Subject: [Freeswitch-users] User Directory and Per-user (Channel) variables Message-ID: bridge() appears to be ignoring the "absolute_codec_string" channel variable defined in the User Directory even though "info" shows that it is present. Other variables, such as "effective_caller_id_number" seem to behave correctly which leads me to believe that this may be a very minor bug. In order to ease trouble shooting, I have tried to implement it using a configuration that clings rather closely to the sample/default configuration files... // User Directory sample // Dialplan broken sample (relies on the channel variable defined in the User Directory) // Dialplan working sample (explicit use of the channel variable) // Gateway sample From brian at freeswitch.org Mon May 18 11:25:02 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 18 May 2009 13:25:02 -0500 Subject: [Freeswitch-users] User Directory and Per-user (Channel) variables In-Reply-To: References: Message-ID: Because by the time it gets here... the codec is already picked.. you'll have to turn on late neg. for this to work. /b On May 18, 2009, at 1:08 PM, Metik wrote: > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/0299d755/attachment-0001.html From freeswitch-users-list at metik.com Mon May 18 12:07:06 2009 From: freeswitch-users-list at metik.com (Metik) Date: Mon, 18 May 2009 15:07:06 -0400 Subject: [Freeswitch-users] User Directory and Per-user (Channel)variables In-Reply-To: References: Message-ID: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> Oddly enough, I initially though that was the problem and enabled it without any success... freeswitch at noesis.metik.com> sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.100 Ext-RTP-IP 192.168.1.100 SIP-IP 192.168.1.100 Ext-SIP-IP 192.168.1.100 URL sip:mod_sofia at 192.168.1.100:5062 BIND-URL sip:mod_sofia at 192.168.1.100:5062;maddr=192.168.1.100 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 5 FAILED-CALLS-IN 0 CALLS-OUT 10 FAILED-CALLS-OUT 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/64c7ac80/attachment.html From brian at freeswitch.org Mon May 18 12:18:03 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 18 May 2009 14:18:03 -0500 Subject: [Freeswitch-users] User Directory and Per-user (Channel)variables In-Reply-To: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> References: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> Message-ID: Are you authenticating phone calls? Also hop on IRC this email ping pong is too slow. /b On May 18, 2009, at 2:07 PM, Metik wrote: > Oddly enough, I initially though that was the problem and enabled it > without any success... Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/79660e84/attachment.html From dujinfang at gmail.com Mon May 18 12:21:58 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 19 May 2009 03:21:58 +0800 Subject: [Freeswitch-users] Logging 503's or other errors In-Reply-To: <191c3a030905180605w31b6114aoceabb866108b2a7@mail.gmail.com> References: <3885f4fe0905171126y7eb9b419x4c103a53cf4a6cad@mail.gmail.com> <191c3a030905180605w31b6114aoceabb866108b2a7@mail.gmail.com> Message-ID: Even the b leg cdr is enabled it only remember the final state(channel vars) on the b leg. At least there are two possible ways to keep tracking all the gateways: 1) don't use '|' separated dial string, use a lua script like this: session:execute("bridge", dial_string1); bridge_hangup_cause = session:getVariable("bridge_hangup_cause") or session:getVariable("originate_disposition"); if (bridge_hangup_cause == "NORMAL_TEMPORARY_FAILURE" or bridge_hangup_cause == "NO_ROUTE_DESTINATION" or bridge_hangup_cause == "CALL_REJECTED") then freeswitch.consoleLog("notice", "Hangup. Cause: [" .. bridge_hangup_cause .. "]. Retry: " -- database.insert('something') session:execue("bridge", dial_string2); if (bridge_hangup_cause == "NORMAL_TEMPORARY_FAILURE" or bridge_hangup_cause == "NO_ROUTE_DESTINATION" or bridge_hangup_cause == "CALL_REJECTED") then session:execute("bridge", dial_string3); .... obviously it can be done in a loop 2) by sip: add a custom header to INVITE, bridge({sip_h_x_xxx=yyy}sofia/gateways/a/...|sofia/gateways/b/...| sofia/gateways/c/... be sure to give yyy a unique value each time you call, then you can dump all the sip messages and by cross reference of the sip_h_x_xxx and call-ID you can get all the related sip messages(every INVITE will have the same sip_h_x_xxx header and each INVITE related message will have the same call-ID. On May 18, 2009, at 9:05 PM, Anthony Minessale wrote: > enable the b leg cdr as well and you will also get cdr from the b > leg perspective. > both xml cdr and cdr csv have params in the config to enable it. > > > On Sun, May 17, 2009 at 1:26 PM, Ron McCarthy > wrote: > Hi list, > > Ive been trying to find a way to log 503's, 480's and other SIP > response codes. If we have continue_on_fail=true and have multiple > gateways for the call to go out, if the 1st,2nd or whatever gateways > fail can we log it somehow? We'd like to know if a carrier is having > issues or not letting us send calls for some reason, from what I can > tell I only show one CDR get written and that's at the end of the > call, so it says nothing about the gateways we tried to send a call > before and if they failed. > > Any ideals on how to do this? Im using the XML CURL dialplan if that > matter. Any ideals how this could be setup so we can keep track of > what is going on? > > Thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/5c2713fa/attachment.html From mrene_lists at avgs.ca Mon May 18 12:25:25 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 18 May 2009 21:25:25 +0200 Subject: [Freeswitch-users] User Directory and Per-user (Channel)variables In-Reply-To: References: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> Message-ID: absolute_codec_string needs to be available from the B-leg too so it can be used on outbound channels. Add that to your directory entry and it should work. Math On 18-May-09, at 9:18 PM, Brian West wrote: > Are you authenticating phone calls? Also hop on IRC this email ping > pong is too slow. > > /b > > On May 18, 2009, at 2:07 PM, Metik wrote: > >> Oddly enough, I initially though that was the problem and enabled >> it without any success... > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/2502f0ff/attachment-0001.html From anthony.minessale at gmail.com Mon May 18 12:29:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 18 May 2009 14:29:03 -0500 Subject: [Freeswitch-users] User Directory and Per-user (Channel)variables In-Reply-To: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> References: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> Message-ID: <191c3a030905181229y7e891934i2a4fc8157aabd8a6@mail.gmail.com> you for sure need late negotiation: also: You are only setting the variable on the inbound leg but not the outbound leg. Remember there are 2 separate channels here. Try this in the same place you are setting the caller id in your broken example: you could also do this to pass it across On Mon, May 18, 2009 at 2:07 PM, Metik wrote: > Oddly enough, I initially though that was the problem and enabled > it without any success... > > freeswitch at noesis.metik.com> sofia status profile internal > API CALL [sofia(status profile internal)] output: > > ================================================================================================= > Name internal > Domain Name N/A > DBName sofia_reg_internal > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_from > RTP-IP 192.168.1.100 > Ext-RTP-IP 192.168.1.100 > SIP-IP 192.168.1.100 > Ext-SIP-IP 192.168.1.100 > URL sip:mod_sofia at 192.168.1.100:5062 > BIND-URL sip:mod_sofia at 192.168.1.100:5062 > ;maddr=192.168.1.100 > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS G722,PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG true > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 5 > FAILED-CALLS-IN 0 > CALLS-OUT 10 > FAILED-CALLS-OUT 0 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/8c1b28fb/attachment.html From dujinfang at gmail.com Mon May 18 12:34:16 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 19 May 2009 03:34:16 +0800 Subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Message-ID: On register, sometimes my voip client got SIP/2.0 482 Request merged sometimes got 200 ok. 482 also means loop detected. my client only has one account logged in only one place, and no proxy, can I take 482 as 200 OK? Thanks. from RFC 3261: "8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction." From brian at freeswitch.org Mon May 18 12:37:24 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 18 May 2009 14:37:24 -0500 Subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK In-Reply-To: References: Message-ID: Is this in regards to FreeSWITCH or something else you're writing? /b On May 18, 2009, at 2:34 PM, dujinfang wrote: > On register, sometimes my voip client got SIP/2.0 482 Request merged > sometimes got 200 ok. > > 482 also means loop detected. my client only has one account logged in > only one place, and no proxy, can I take 482 as 200 OK? > > Thanks. > > from RFC 3261: > > "8.2.2.2 Merged Requests > > If the request has no tag in the To header field, the UAS core MUST > check the request against ongoing transactions. If the From tag, > Call-ID, and CSeq exactly match those associated with an ongoing > transaction, but the request does not match that transaction (based > on the matching rules in Section 17.2.3), the UAS core SHOULD > generate a 482 (Loop Detected) response and pass it to the server > transaction." > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/1e6e779d/attachment.html From anthony.minessale at gmail.com Mon May 18 12:37:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 18 May 2009 14:37:05 -0500 Subject: [Freeswitch-users] User Directory and Per-user (Channel)variables In-Reply-To: References: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> Message-ID: <191c3a030905181237q73011972s3c98c62408e0dacd@mail.gmail.com> =D That's another way I didn't mention. There are 2 more but they are more complicated so I will omit them ;) On Mon, May 18, 2009 at 2:25 PM, Mathieu Rene wrote: > absolute_codec_string needs to be available from the B-leg too so it can be > used on outbound channels. > Add that to your directory entry and it should work. > > > > Math > > On 18-May-09, at 9:18 PM, Brian West wrote: > > Are you authenticating phone calls? Also hop on IRC this email ping pong > is too slow. > /b > > On May 18, 2009, at 2:07 PM, Metik wrote: > > Oddly enough, I initially though that was the problem and enabled > it without any success... > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/7268ed2a/attachment-0001.html From freeswitch-users-list at metik.com Mon May 18 12:43:43 2009 From: freeswitch-users-list at metik.com (Metik) Date: Mon, 18 May 2009 15:43:43 -0400 Subject: [Freeswitch-users] User Directory and Per-user(Channel)variables In-Reply-To: References: <9A1A14D98D2C4A61B4D651B33A48D044@chaos> Message-ID: <35B51E27B93A488FBF80AA55A8A88629@chaos> Math, That was it--thank you very much! -Metik ----- Original Message ----- From: Mathieu Rene To: freeswitch-users at lists.freeswitch.org Sent: Monday, May 18, 2009 3:25 PM Subject: Re: [Freeswitch-users] User Directory and Per-user(Channel)variables absolute_codec_string needs to be available from the B-leg too so it can be used on outbound channels. Add that to your directory entry and it should work. Math On 18-May-09, at 9:18 PM, Brian West wrote: Are you authenticating phone calls? Also hop on IRC this email ping pong is too slow. /b On May 18, 2009, at 2:07 PM, Metik wrote: Oddly enough, I initially though that was the problem and enabled it without any success... Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/fe9a3ddc/attachment.html From dujinfang at gmail.com Mon May 18 16:55:40 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 19 May 2009 07:55:40 +0800 Subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK In-Reply-To: References: Message-ID: Yes, FS(13263) send out 482 request merged to my voip client. I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below : recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:32.811280: REGISTER sip:voip.xxx.com SIP/2.0 CSeq: 208 REGISTER Content-Length: 0 recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:38.814237: REGISTER sip:voip.xxx.com SIP/2.0 CSeq: 210 REGISTER recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:48.821027: REGISTER sip:voip.xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKa9b7ba70783b617e9998dc4dd82eb3c5 From: ;tag=a9b7ba70783b617e9998dc4dd82eb3c5 To: "cc" Call-ID: a9b7ba70783b617e9998dc4dd82eb3c5 at 192.168.1.100 CSeq: 214 REGISTER Contact: max-forwards: 70 expires: 300 Content-Length: 0 recv 629 bytes from udp/[69.131.94.250]:3270 at 12:40:52.841591: REGISTER sip:voip.xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da From: "cc" ;tag=b8c37e33defde51cf91e1e03e51657da To: "cc" Call-ID: b8c37e33defde51cf91e1e03e51657da at 192.168.1.100 CSeq: 1 REGISTER Contact: max-forwards: 70 expires: 300 sent 439 bytes to udp/[69.131.94.250]:3270 at 12:40:52.841767: SIP/2.0 482 Request merged Via: SIP/2.0/UDP 192.168.1.100 : 3270 ;rport = 3270 ;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da;received=69.131.94.250 From: "cc" ;tag=b8c37e33defde51cf91e1e03e51657da To: "cc" ;tag=tjgccmtraDHFc Call-ID: b8c37e33defde51cf91e1e03e51657da at 192.168.1.100 CSeq: 1 REGISTER Content-Length: 0 And I also noticed the CSeq if not continues, seems it lost some. but why the CSeq so big while the client directly logins to FS without any proxy and I don't think there is a loop? Anyway, don't know why FS does not respond to REGISTER sometimes. I updated FS to 13374, will see if it happen again. On May 19, 2009, at 3:37 AM, Brian West wrote: > Is this in regards to FreeSWITCH or something else you're writing? > > /b > > On May 18, 2009, at 2:34 PM, dujinfang wrote: > >> On register, sometimes my voip client got SIP/2.0 482 Request merged >> sometimes got 200 ok. >> >> 482 also means loop detected. my client only has one account logged >> in >> only one place, and no proxy, can I take 482 as 200 OK? >> >> Thanks. >> >> from RFC 3261: >> >> "8.2.2.2 Merged Requests >> >> If the request has no tag in the To header field, the UAS core >> MUST >> check the request against ongoing transactions. If the From tag, >> Call-ID, and CSeq exactly match those associated with an ongoing >> transaction, but the request does not match that transaction >> (based >> on the matching rules in Section 17.2.3), the UAS core SHOULD >> generate a 482 (Loop Detected) response and pass it to the server >> transaction." >> > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/fdc6d2ba/attachment.html From brian at freeswitch.org Mon May 18 17:06:09 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 18 May 2009 19:06:09 -0500 Subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK In-Reply-To: References: Message-ID: <021CCE0C-19A4-43D1-A081-E5A551BA78F3@freeswitch.org> Please show me a pcap file to may email address because I can bet you FS/Sofia is doing it right 99% of the time. /b On May 18, 2009, at 6:55 PM, dujinfang wrote: > es, FS(13263) send out 482 request merged to my voip client. > > I guess, for some reason, FS doesn't respond to the REGISTER, and > when the client start REGISTER again using another call-id, it > merged the request to one. Anyone ever met this before? See the call- > id and cseq below : > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/8de85c53/attachment-0001.html From dujinfang at gmail.com Mon May 18 17:39:04 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 19 May 2009 08:39:04 +0800 Subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK In-Reply-To: <021CCE0C-19A4-43D1-A081-E5A551BA78F3@freeswitch.org> References: <021CCE0C-19A4-43D1-A081-E5A551BA78F3@freeswitch.org> Message-ID: <21C44423-90A8-4A6E-A4CF-09B759985556@gmail.com> I believe. capturing on tshark -i eth1 -w register.pcap udp port 5090 do you have further suggestions on the tshark filter? Thanks. On May 19, 2009, at 8:06 AM, Brian West wrote: > Please show me a pcap file to may email address because I can bet > you FS/Sofia is doing it right 99% of the time. > > /b > > On May 18, 2009, at 6:55 PM, dujinfang wrote: > >> es, FS(13263) send out 482 request merged to my voip client. >> >> I guess, for some reason, FS doesn't respond to the REGISTER, and >> when the client start REGISTER again using another call-id, it >> merged the request to one. Anyone ever met this before? See the >> call-id and cseq below : >> > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/5dce85c4/attachment.html From brian at freeswitch.org Mon May 18 17:42:32 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 18 May 2009 19:42:32 -0500 Subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK In-Reply-To: <21C44423-90A8-4A6E-A4CF-09B759985556@gmail.com> References: <021CCE0C-19A4-43D1-A081-E5A551BA78F3@freeswitch.org> <21C44423-90A8-4A6E-A4CF-09B759985556@gmail.com> Message-ID: <28FEEB8D-9535-42C1-8D9C-21F28E7F3B91@freeswitch.org> I never use tshark to capture... I use tcpdump -s0 -x port 5090 -w file.pcap /b On May 18, 2009, at 7:39 PM, dujinfang wrote: > I believe. > > capturing on tshark -i eth1 -w register.pcap udp port 5090 > > do you have further suggestions on the tshark filter? > > Thanks. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090518/e1fc68ff/attachment.html From pawzlion at gmail.com Mon May 18 18:02:50 2009 From: pawzlion at gmail.com (David Robinson) Date: Tue, 19 May 2009 11:02:50 +1000 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user In-Reply-To: References: Message-ID: <651FE195-03AC-4D51-957B-B6EF463939AB@gmail.com> > My suspicion is that the RTP traffic isn't traversing the NAT > properly. You > may have to configure the routers at both ends to forward the RTP > packets to > the correct destinations. There is a good discussion of NAT on the > wiki. Situation: FS (10.0.0.12) -> DMZ (124.254.81.250) -> Internet -> NAT (203.206.171.118) -> Softphone (10.0.0.2) The problem is there's so much discussion of NAT that I'm not sure where to start. OK the problem is that I can't control the "external" user's router so I need a solution that works by only fixing the FS end. I've put my FS in the DMZ, but of course it's still got a local LAN IP address. Is there something I can configure to make FS realise that it _doesn't_ need to use NAT ? Whenever my softphones register to FS they register as UDP-NAT. Can I prevent that and make them register as regular UDP ? It would seem like they don't need to be in NAT mode since FS is in a DMZ, or do they ? I tried setting inbound-late-negotiation in my external (is this right ?) SIP profile and added proxy_media to my extension configurations in the dialplans, but this made no difference. It's possible that I haven't done this in the right spot or something. The other thing that looks promising is on http://wiki.freeswitch.org/wiki/External_profile which gives an example of a softphone registering to a NAT'd FS from outside on the internet (Switch with External Softphone example) which suggests I create a new external profile on a different port. I've done this and the user's softphone can register fine, but when he makes calls we still get no audio, presumably from lack of RTP data. I then tried adding in values for rtp-ip, sip-ip, ext-rtp-ip and ext-sip- ip on the new external profile to see if that made any difference but it didn't. Step 6 of the example says "reference the caller from your FreeSWITCH system as: sofia/external5090/@x.x.x.x: 5090". I'm not sure what that means. Do I have to change something else to make it "reference" the caller by that external profile ? I figured it must be at least using that external profile because the phone is successfully registering on port 5090, but I'm not sure if I have to do something different to route incoming calls from the main external profile to the new 5090 one. I'm just not sure which NAT-related solution I'm supposed to be using. The External_profile wiki page example for the external softphone seems to fit my situation but didn't solve anything. The proxy_media solution seemed promising but had no real effect. It seems to me that the solution has something to do with having FS know that it's in a DMZ and that it doesn't need to do any NAT traversal, thereby making it think it's got a live internet IP and therefore only the external user would be using NAT traversal. I hope someone can give me some insight into which particular NAT- related solution I need because there seems to be dozens of ways to deal with this problem and I can't figure out which applies. From adam.falcone at gmail.com Mon May 18 20:39:43 2009 From: adam.falcone at gmail.com (adamF) Date: Mon, 18 May 2009 20:39:43 -0700 (PDT) Subject: [Freeswitch-users] No ringback for iPhone Message-ID: <23609250.post@talk.nabble.com> I am not receiving any ringback when calling in from an iPhone. I receive ringback when calling from other cell phones and land lines just not an iPhone. 2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/3153836680 at 64.24.35.76 Execute set(ringback=${us-ring}) 2009-05-18 20:28:04 [DEBUG] switch_core_session.c:1286 switch_core_session_exec() sofia/external/3153836680 at 64.24.35.76 Expanded String set(ringback=%(2000, 4000, 440.0, 480.0)) 2009-05-18 20:28:04 [DEBUG] mod_dptools.c:711 set_function() sofia/external/3153836680 at 64.24.35.76 SET [ringback]=[%(2000, 4000, 440.0, 480.0)] 2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/3153836680 at 64.24.35.76 Execute set(transfer_ringback=local_stream://moh) 2009-05-18 20:28:04 [DEBUG] mod_dptools.c:711 set_function() sofia/external/3153836680 at 64.24.35.76 SET [transfer_ringback]=[local_stream://moh] Any suggestions? -- View this message in context: http://www.nabble.com/No-ringback-for-iPhone-tp23609250p23609250.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From saeedahmad1981 at gmail.com Tue May 19 00:40:26 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 19 May 2009 09:40:26 +0200 Subject: [Freeswitch-users] text to speech IVRs and MOH Message-ID: Hi all, Could you guys recommend me any online text to speech IVR software which works OK with FS. i am using AT&T site and for some IVRs i get sample rate errors. Also some resource to download more MOH wav files. Many thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/57b86722/attachment.html From kjv at ken-ton.com Tue May 19 04:59:30 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Tue, 19 May 2009 07:59:30 -0400 Subject: [Freeswitch-users] Cisco 7905G IP Phone and Freeswitch In-Reply-To: <0D04867F-78FE-4A23-A3C9-1A63D840F0B4@freeswitch.org> References: <771c87020905151447td7e475albe7b00b480e552d4@mail.gmail.com> <0D04867F-78FE-4A23-A3C9-1A63D840F0B4@freeswitch.org> Message-ID: <926E2CA2-341D-4452-868E-1AB0007C1CC4@ken-ton.com> Brian; We went over this... http://wiki.freeswitch.org/wiki/Freeswitch_Cisco_7960G_Howto He's got exactly the same problem I had... It's addressed in the wiki. Follow the destructions exactly... Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On May 15, 2009, at 6:19 PM, Brian West wrote: > What port do these cisco's register with? > > /b > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/9036585b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/9036585b/attachment-0001.bin From anthony.minessale at gmail.com Tue May 19 05:46:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 May 2009 07:46:54 -0500 Subject: [Freeswitch-users] No ringback for iPhone In-Reply-To: <23609250.post@talk.nabble.com> References: <23609250.post@talk.nabble.com> Message-ID: <191c3a030905190546j4c42c4bas2e14ec93e09b0719@mail.gmail.com> maybe ATT doesn't do early media on their network. you could probably answer the call before you bridge if the ringback is just for aesthetics. On Mon, May 18, 2009 at 10:39 PM, adamF wrote: > > I am not receiving any ringback when calling in from an iPhone. I receive > ringback when calling from other cell phones and land lines just not an > iPhone. > > > 2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/external/3153836680 at 64.24.35.76 > Execute set(ringback=${us-ring}) > 2009-05-18 20:28:04 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/external/3153836680 at 64.24.35.76 Expanded > String set(ringback=%(2000, 4000, 440.0, 480.0)) > 2009-05-18 20:28:04 [DEBUG] mod_dptools.c:711 set_function() > sofia/external/3153836680 at 64.24.35.76 SET [ringback]=[%(2000, 4000, 440.0, > 480.0)] > 2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/external/3153836680 at 64.24.35.76 > Execute set(transfer_ringback=local_stream://moh) > 2009-05-18 20:28:04 [DEBUG] mod_dptools.c:711 set_function() > sofia/external/3153836680 at 64.24.35.76 SET > [transfer_ringback]=[local_stream://moh] > > Any suggestions? > -- > View this message in context: > http://www.nabble.com/No-ringback-for-iPhone-tp23609250p23609250.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/c85d045f/attachment.html From maxim.tsvetov at gmail.com Tue May 19 00:49:27 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Tue, 19 May 2009 00:49:27 -0700 (PDT) Subject: [Freeswitch-users] Testimonials Message-ID: <23611131.post@talk.nabble.com> Hello, Our company want to use Freeswitch and now we testing this solution in the lab. If someone already using Freeswitch as office pbx, ivr or any other commercial purposes could you please share you experience (and if it is not a secret - company name)? Regards, Maxim Tsvetov -- View this message in context: http://www.nabble.com/Testimonials-tp23611131p23611131.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From rupa at rupa.com Tue May 19 05:53:12 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 19 May 2009 07:53:12 -0500 Subject: [Freeswitch-users] No ringback for iPhone In-Reply-To: <191c3a030905190546j4c42c4bas2e14ec93e09b0719@mail.gmail.com> References: <23609250.post@talk.nabble.com> <191c3a030905190546j4c42c4bas2e14ec93e09b0719@mail.gmail.com> Message-ID: On Tue, May 19, 2009 at 7:46 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > maybe ATT doesn't do early media on their network. > you could probably answer the call before you bridge if the ringback is > just for aesthetics. > > ATT definitely does early media on their network (at least in my market -- Dallas). I get early media when falling FS from my iphone. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/50601b9d/attachment.html From anthony.minessale at gmail.com Tue May 19 05:56:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 May 2009 07:56:36 -0500 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user In-Reply-To: <651FE195-03AC-4D51-957B-B6EF463939AB@gmail.com> References: <651FE195-03AC-4D51-957B-B6EF463939AB@gmail.com> Message-ID: <191c3a030905190556l8ffd561h19bafa90a5369b9c@mail.gmail.com> edit your sip profile and comment out every line that contains the string nat to disable all the nat auto-detection. for dmz, you need to set the rtp-ext-ip and sip-ext-ip to be the live ip and sip-ip and rtp-ip to be the lan ip (the real one) On Mon, May 18, 2009 at 8:02 PM, David Robinson wrote: > > My suspicion is that the RTP traffic isn't traversing the NAT > > properly. You > > may have to configure the routers at both ends to forward the RTP > > packets to > > the correct destinations. There is a good discussion of NAT on the > > wiki. > > > Situation: FS (10.0.0.12) -> DMZ (124.254.81.250) -> Internet -> NAT > (203.206.171.118) -> Softphone (10.0.0.2) > > The problem is there's so much discussion of NAT that I'm not sure > where to start. OK the problem is that I can't control the "external" > user's router so I need a solution that works by only fixing the FS > end. I've put my FS in the DMZ, but of course it's still got a local > LAN IP address. Is there something I can configure to make FS realise > that it _doesn't_ need to use NAT ? Whenever my softphones register to > FS they register as UDP-NAT. Can I prevent that and make them register > as regular UDP ? It would seem like they don't need to be in NAT mode > since FS is in a DMZ, or do they ? > > I tried setting inbound-late-negotiation in my external (is this > right ?) SIP profile and added proxy_media to my extension > configurations in the dialplans, but this made no difference. It's > possible that I haven't done this in the right spot or something. > > The other thing that looks promising is on > http://wiki.freeswitch.org/wiki/External_profile > which gives an example of a softphone registering to a NAT'd FS from > outside on the internet (Switch with External Softphone example) which > suggests I create a new external profile on a different port. I've > done this and the user's softphone can register fine, but when he > makes calls we still get no audio, presumably from lack of RTP data. I > then tried adding in values for rtp-ip, sip-ip, ext-rtp-ip and ext-sip- > ip on the new external profile to see if that made any difference but > it didn't. Step 6 of the example says "reference the caller from your > FreeSWITCH system as: sofia/external5090/@x.x.x.x: > 5090". I'm not sure what that means. Do I have to change something > else to make it "reference" the caller by that external profile ? I > figured it must be at least using that external profile because the > phone is successfully registering on port 5090, but I'm not sure if I > have to do something different to route incoming calls from the main > external profile to the new 5090 one. > > I'm just not sure which NAT-related solution I'm supposed to be using. > The External_profile wiki page example for the external softphone > seems to fit my situation but didn't solve anything. The proxy_media > solution seemed promising but had no real effect. It seems to me that > the solution has something to do with having FS know that it's in a > DMZ and that it doesn't need to do any NAT traversal, thereby making > it think it's got a live internet IP and therefore only the external > user would be using NAT traversal. > > I hope someone can give me some insight into which particular NAT- > related solution I need because there seems to be dozens of ways to > deal with this problem and I can't figure out which applies. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/97bc7ac7/attachment.html From brian at freeswitch.org Tue May 19 06:24:27 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 08:24:27 -0500 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user In-Reply-To: <191c3a030905190556l8ffd561h19bafa90a5369b9c@mail.gmail.com> References: <651FE195-03AC-4D51-957B-B6EF463939AB@gmail.com> <191c3a030905190556l8ffd561h19bafa90a5369b9c@mail.gmail.com> Message-ID: <27414EDF-1B3C-4453-962F-0C6CAD1D46F3@freeswitch.org> You will also need to modify the dial-string in conf/directory/ default.xml because it only looks on internal for registered users. /b On May 19, 2009, at 7:56 AM, Anthony Minessale wrote: > edit your sip profile and comment out every line that contains the > string nat to disable all the nat auto-detection. > for dmz, you need to set the rtp-ext-ip and sip-ext-ip to be the > live ip and sip-ip and rtp-ip to be the lan ip (the real one) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/a4dd9477/attachment-0001.html From brian at freeswitch.org Tue May 19 06:25:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 08:25:56 -0500 Subject: [Freeswitch-users] No ringback for iPhone In-Reply-To: References: <23609250.post@talk.nabble.com> <191c3a030905190546j4c42c4bas2e14ec93e09b0719@mail.gmail.com> Message-ID: <41126BB5-8212-4669-93F7-3D6898C5D527@freeswitch.org> I have heard mine not provide ringback a few times... heck I have had mine ring but not actually RING my iphone :P GO AT&T, you're so freakin awesome! /b On May 19, 2009, at 7:53 AM, Rupa Schomaker wrote: > > ATT definitely does early media on their network (at least in my > market -- Dallas). I get early media when falling FS from my iphone. > > -- > -Rupa > ________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From klejch+freeswitch at netbox.cz Tue May 19 07:12:00 2009 From: klejch+freeswitch at netbox.cz (Vladimir Klejch) Date: Tue, 19 May 2009 16:12:00 +0200 (CEST) Subject: [Freeswitch-users] Problem with XML preprocessor variable Message-ID: Hi I think i found a bug on XML preprocessor in way of working with preprocessor varibles. I have problem with concatenation of preprocessor variables in XML config. Preprocessor is incorectly concatenatin two preprocessor variables, and building wrong config xml. Best to explain on example: in vars.xml: then in dialplan: but in freeswitch.xml.fsxml: The odd variables are wrong translated by preprocessor . The odd variable used as $${var} is translated corect to his content, but the even variable used as $${var} is mangled and in freeswitch.xml.fsxml is to seen %${var} form of prefious $${var}. This problem only ocur, if tvo variables are direcly concatenated w/o any char between them. config : is translated to string: "xpreprocessbug1$%{bug_test_2}xpreprocessbug3$%{bug_test_4}" , to see in extension name="x_preprocess_bug1". If there is a char between concatenated variables (in my test a ^ char), everything is OK, to see in extension name="x_preprocess_bug2". It's this known problem or I'm wrong understanding preprocessor usage ?? Thanks Kleo -- kleo+freeswitch at netbox.cz From anthony.minessale at gmail.com Tue May 19 08:39:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 May 2009 10:39:53 -0500 Subject: [Freeswitch-users] Problem with XML preprocessor variable In-Reply-To: References: Message-ID: <191c3a030905190839q18522e13tf9c1b3280ec3a460@mail.gmail.com> yes it was a bug, it has been fixed in r13385 please report future bugs to http://jira.freeswitch.org not the mailing list. On Tue, May 19, 2009 at 9:12 AM, Vladimir Klejch < klejch+freeswitch at netbox.cz > wrote: > > Hi > > I think i found a bug on XML preprocessor in way of working with > preprocessor varibles. > > I have problem with concatenation of preprocessor variables in XML config. > > Preprocessor is incorectly concatenatin two preprocessor variables, and > building wrong config xml. > > Best to explain on example: > > in vars.xml: > > > > > > > > > data="bug_test_sum1=$${bug_test_1}$${bug_test_2}$${bug_test_3}$${bug_test_4}" > /> > data="bug_test_sum2=$${bug_test_1}^$${bug_test_2}^$${bug_test_3}^$${bug_test_4}" > /> > > > then in dialplan: > > > > expression="$${bug_test_sum1}" > > > > > > > expression="$${bug_test_sum2}" > > > > > > > > > but in freeswitch.xml.fsxml: > > > expression="xpreprocessbug1$%{bug_test_2}xpreprocessbug3$%{bug_test_4}" > > > > > > > expression="xpreprocessbug1^xpreprocessbug2^xpreprocessbug3^xpreprocessbug4" > > > > > > > > > The odd variables are wrong translated by preprocessor . The odd variable > used as $${var} is translated corect to his content, but the even variable > used as $${var} is mangled and in freeswitch.xml.fsxml is to seen %${var} > form of prefious $${var}. This problem only ocur, if tvo variables are > direcly concatenated w/o any char between them. > > config : > > data="bug_test_sum1=$${bug_test_1}$${bug_test_2}$${bug_test_3}$${bug_test_4}" > /> > > is translated to string: > > "xpreprocessbug1$%{bug_test_2}xpreprocessbug3$%{bug_test_4}" , to see in > extension name="x_preprocess_bug1". > > If there is a char between concatenated variables (in my test a ^ char), > everything is OK, to see in extension name="x_preprocess_bug2". > > > > It's this known problem or I'm wrong understanding preprocessor usage ?? > > > > > Thanks > Kleo > > -- > kleo+freeswitch at netbox.cz > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/fb01d041/attachment.html From pete at privateconnect.com Tue May 19 09:31:19 2009 From: pete at privateconnect.com (pete at privateconnect.com) Date: Tue, 19 May 2009 09:31:19 -0700 Subject: [Freeswitch-users] text to speech IVRs and MOH Message-ID: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/a37f568d/attachment.html From dave at 3c.co.uk Tue May 19 10:21:08 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 19 May 2009 18:21:08 +0100 Subject: [Freeswitch-users] text to speech IVRs and MOH In-Reply-To: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net> References: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net> Message-ID: <1242753668.20213.3.camel@dk-d820> Just to add one: Cepstral - quite a lot cheaper than the non-free ones Pete mentioned; voice quality quite adequate for what we want it for, which is short prompts and development. Not tried it for reading e-mail or the like. FS' mod_cepstral isn't wholly compatible with their 5.x release (unless someone's fixed it), but that's easily worked around. --Dave > I've spent the last 2-3 months on researching TTS and ASR for FS for a > project. Best TTS depends on what you consider important. Also, how > do you plan on using it. > > > Here's some of the TTS engines I've run across with some pros/cons: > > > Festivate Lite (flite) > Pros: > - Free (comes with FS) > - simple to use > - 16K voice sounds decent > - Completely customizable > Cons: > - 8K voice sounds horrible over cell phone > > > NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl > voice) > Pros: > - My selection for best soundig voices > - Recently select by Stephen Hawkings for his voice (geek points!) > - Lots of Languages supported > - Free trial available > Cons: > - Custom C-Based API (FS interface coming soon) > - Large file size (Engine + SDK + 1 Voice = 900MB) > - Support is lacking (Company beed in Korean, time zone issues, etc) > > > Nuance ($500/port for 1 voice) > Pros: > - Wide Variety of Products > - Support MRCP > - Supports ASR as well (add'l fees) > - Excellent support > - Free trial > - Decent sounding voices at 8K and 16K > - Wide range of tuning parameters > Cons: > - Pricey > - Limited voice selection > - Limited support for 64-bit linux > > > AT&T (NaturalVoice) (no pricing info available) > Pros: > - Big company (solid in marketplace) > - Good suppport (user and developer) > - ASP model means no software to maintain > Cons: > - ASP model incurs delay > - Voices sound too digitized > - Limited support for 64-bit linux > > > > Loquendo ($500/port for 1 voice + 15% addl voice) > Pros: > - Good sounding voices (almost as good as NeoSpeech) > - Wide variety of languages > - Excellent support > - Has free 30 day trial > - Supports MRCP > - Support ASR and Voice Recognition as well. (add'l fees) > - Small footprint (< 150MB) > Cons: > - Pricey > - Complicated install process > - Limited management/tuning capabilities > > > In the end, it was down to NeoSpeech or Loquendo for our application. > We are currently running tests with NeoSpeech and assuming all goes > well, we will select them. Though don't let that color your opinion > too much after several "focus groups" we discovered the most important > element in the equation is does your customer/boss like the sound of > the voices, and that is a completely subjective decision. > > > > > -pete > > > -------- Original Message -------- > Subject: [Freeswitch-users] text to speech IVRs and MOH > From: Saeed Ahmad > Date: Tue, May 19, 2009 12:40 am > To: freeswitch-users at lists.freeswitch.org > > Hi all, > > Could you guys recommend me any online text to speech IVR > software which works OK with FS. i am using AT&T site and for > some IVRs i get sample rate errors. Also some resource to > download more MOH wav files. > > Many thanks > > ______________________________________________________________ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From anthony.minessale at gmail.com Tue May 19 10:28:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 May 2009 12:28:15 -0500 Subject: [Freeswitch-users] As seen on DIGG Message-ID: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> please find it in your hearts to digg this story. http://digg.com/search?s=skype+switch -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/92639beb/attachment.html From dave at 3c.co.uk Tue May 19 10:29:33 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 19 May 2009 18:29:33 +0100 Subject: [Freeswitch-users] Testimonials In-Reply-To: <23611131.post@talk.nabble.com> References: <23611131.post@talk.nabble.com> Message-ID: <1242754173.20213.5.camel@dk-d820> Hi Maxim - We've used FreeSWITCH for switching large volumes of wholesale traffic and for a variety of IVR services; we no longer use anything else. See http://www.softivr.com for something which we've built on it. Cheers -- Dave > Hello, > > Our company want to use Freeswitch and now we testing this solution in the > lab. > > If someone already using Freeswitch as office pbx, ivr or any other > commercial purposes > could you please share you experience (and if it is not a secret - company > name)? > > Regards, > Maxim Tsvetov -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From Prometheus001 at gmx.net Tue May 19 10:31:23 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 19 May 2009 19:31:23 +0200 Subject: [Freeswitch-users] text to speech IVRs and MOH In-Reply-To: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net> References: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net> Message-ID: <4A12ECEB.1050207@gmx.net> Thanks for this overwiev. One question: How does this compare to Cepstral TTS? Best regards Peter pete at privateconnect.com schrieb: > I've spent the last 2-3 months on researching TTS and ASR for FS for a > project. Best TTS depends on what you consider important. Also, how > do you plan on using it. > > Here's some of the TTS engines I've run across with some pros/cons: > > Festivate Lite (flite) > Pros: > - Free (comes with FS) > - simple to use > - 16K voice sounds decent > - Completely customizable > Cons: > - 8K voice sounds horrible over cell phone > > NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl voice) > Pros: > - My selection for best soundig voices > - Recently select by Stephen Hawkings for his voice (geek points!) > - Lots of Languages supported > - Free trial available > Cons: > - Custom C-Based API (FS interface coming soon) > - Large file size (Engine + SDK + 1 Voice = 900MB) > - Support is lacking (Company beed in Korean, time zone issues, etc) > > Nuance ($500/port for 1 voice) > Pros: > - Wide Variety of Products > - Support MRCP > - Supports ASR as well (add'l fees) > - Excellent support > - Free trial > - Decent sounding voices at 8K and 16K > - Wide range of tuning parameters > Cons: > - Pricey > - Limited voice selection > - Limited support for 64-bit linux > > AT&T (NaturalVoice) (no pricing info available) > Pros: > - Big company (solid in marketplace) > - Good suppport (user and developer) > - ASP model means no software to maintain > Cons: > - ASP model incurs delay > - Voices sound too digitized > - Limited support for 64-bit linux > > Loquendo ($500/port for 1 voice + 15% addl voice) > Pros: > - Good sounding voices (almost as good as NeoSpeech) > - Wide variety of languages > - Excellent support > - Has free 30 day trial > - Supports MRCP > - Support ASR and Voice Recognition as well. (add'l fees) > - Small footprint (< 150MB) > Cons: > - Pricey > - Complicated install process > - Limited management/tuning capabilities > > In the end, it was down to NeoSpeech or Loquendo for our application. > We are currently running tests with NeoSpeech and assuming all goes > well, we will select them. Though don't let that color your opinion > too much after several "focus groups" we discovered the most important > element in the equation is does your customer/boss like the sound of > the voices, and that is a completely subjective decision. > > > -pete > > -------- Original Message -------- > Subject: [Freeswitch-users] text to speech IVRs and MOH > From: Saeed Ahmad > Date: Tue, May 19, 2009 12:40 am > To: freeswitch-users at lists.freeswitch.org > > Hi all, > > Could you guys recommend me any online text to speech IVR software > which works OK with FS. i am using AT&T site and for some IVRs i > get sample rate errors. Also some resource to download more MOH > wav files. > > Many thanks > ------------------------------------------------------------------------ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From maxim.tsvetov at gmail.com Tue May 19 10:53:09 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Tue, 19 May 2009 10:53:09 -0700 (PDT) Subject: [Freeswitch-users] Testimonials In-Reply-To: <1242754173.20213.5.camel@dk-d820> References: <23611131.post@talk.nabble.com> <1242754173.20213.5.camel@dk-d820> Message-ID: <23621332.post@talk.nabble.com> David, thank you for information! What is your experience of this solution - was it easy to support it and to develop new services? What was the highest load on your solution (in number of calls)? David Knell wrote: > > Hi Maxim - > > We've used FreeSWITCH for switching large volumes of wholesale traffic > and for a variety of IVR services; we no longer use anything else. See > http://www.softivr.com for something which we've built on it. > > Cheers -- > > Dave > >> Hello, >> >> Our company want to use Freeswitch and now we testing this solution in >> the >> lab. >> >> If someone already using Freeswitch as office pbx, ivr or any other >> commercial purposes >> could you please share you experience (and if it is not a secret - >> company >> name)? >> >> Regards, >> Maxim Tsvetov > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Testimonials-tp23611131p23621332.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From andrew at hijacked.us Tue May 19 10:54:46 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 19 May 2009 13:54:46 -0400 Subject: [Freeswitch-users] Testimonials In-Reply-To: <23611131.post@talk.nabble.com> References: <23611131.post@talk.nabble.com> Message-ID: <20090519175445.GA25829@hijacked.us> On Tue, May 19, 2009 at 12:49:27AM -0700, Maxim Tsvetov wrote: > > Hello, > > Our company want to use Freeswitch and now we testing this solution in the > lab. > > If someone already using Freeswitch as office pbx, ivr or any other > commercial purposes > could you please share you experience (and if it is not a secret - company > name)? > As of tonight we'll be doing all our PRI<->VoIP gateways using FreeSWITCH and once that's done we'll be replacing our ancient Nortel PBX using FreeSWITCH and probably snom phones. In addition, we're developing a complete open-source call center solution on top of FreeSWITCH (unified communications, skill based routing, all that jazz). I don't know if I should give out my company's name but I can link you to opencsm.org where the callcenter stuff is being developed under a subsidiary company; SpiceCSM. Really, I couldn't ask for anything more than FreeSWITCH has provided, it's the perfect platform for building VoIP enabled systems on top of because at heart it's just a softswitch with lots of cool modules that you can choose if you want or not. Developing for it (writing modules or stuff on top of it) is also great because the code is nice and clean and the interfaces for communicating with FS (xml curl, event socket, etc) are consistant and stable. Andrew From lcm at marshap.com Tue May 19 11:48:20 2009 From: lcm at marshap.com (Larry Marshall) Date: Tue, 19 May 2009 11:48:20 -0700 Subject: [Freeswitch-users] Dialplan bridge action Message-ID: <00c901c9d8b2$615ddf10$24199d30$@com> What are the differences between these two actions? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/e5c341c5/attachment.html From dave at 3c.co.uk Tue May 19 11:54:51 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 19 May 2009 19:54:51 +0100 Subject: [Freeswitch-users] Testimonials In-Reply-To: <23621332.post@talk.nabble.com> References: <23611131.post@talk.nabble.com> <1242754173.20213.5.camel@dk-d820> <23621332.post@talk.nabble.com> Message-ID: <1242759291.20213.14.camel@dk-d820> Hi Maxim, > David, thank you for information! > > What is your experience of this solution - was it easy to support it and to > develop new services? There was a reasonably steep learning curve but, once that's been topped, it's a walk in the park to develop new services. Have a look at some SoftIVR samples: http://www.softivr.com/wiki/index.php/Howtos - by and large, each Javascript function there is implemented using a few (or, at most, a few tens of) lines of Perl which talk to FS over the event socket interface. There's easier ways to get started - FS has a number of built-in scripting languages. > What was the highest load on your solution (in number of calls)? Over 400 bridged calls per box with media (that's 800 call legs, or about 80Mbits/sec of RTP in and out using - as we do - G.711) - other figures that we've reached are 130K calls/day/box and $500k/month/box in turnover :-) Cheers -- Dave > > > David Knell wrote: > > > > Hi Maxim - > > > > We've used FreeSWITCH for switching large volumes of wholesale traffic > > and for a variety of IVR services; we no longer use anything else. See > > http://www.softivr.com for something which we've built on it. > > > > Cheers -- > > > > Dave > > > >> Hello, > >> > >> Our company want to use Freeswitch and now we testing this solution in > >> the > >> lab. > >> > >> If someone already using Freeswitch as office pbx, ivr or any other > >> commercial purposes > >> could you please share you experience (and if it is not a secret - > >> company > >> name)? > >> > >> Regards, > >> Maxim Tsvetov > > -- > > David Knell, Director, 3C Limited > > T: +44 20 3298 2000 > > E: dave at 3c.co.uk > > W: http://www.3c.co.uk > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From brian at freeswitch.org Tue May 19 11:55:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 13:55:32 -0500 Subject: [Freeswitch-users] Dialplan bridge action In-Reply-To: <00c901c9d8b2$615ddf10$24199d30$@com> References: <00c901c9d8b2$615ddf10$24199d30$@com> Message-ID: <0DD2F1D3-1047-4013-91F5-D969ED968780@freeswitch.org> The user/ channel uses the dial-string param on the domain or the user to resolve the correct target to dial. So its like a pseudo channel... Which in the default configs it calls the sofia_contact api to get to the correct user. In your example below with the default config they will not differ in who to ring but in the default I set a variables on the originate line for presence. Example user/1001@$${domain} will expand this: dialed_user and dialed_domain are variables set inside the user/ endpoint when you call it. /b On May 19, 2009, at 1:48 PM, Larry Marshall wrote: > What are the differences between these two actions? > > > > > Thanks, Lars Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/3cfefcc1/attachment-0001.html From niall.crosby at gmail.com Tue May 19 12:21:12 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Tue, 19 May 2009 20:21:12 +0100 Subject: [Freeswitch-users] Load test issue Message-ID: <4aec92830905191221m4e734f39wa3c2ef395bb99310@mail.gmail.com> Hi, Am running SIPP on a pretty fresh default install of FS and am only getting 3 calls processed per second when put under load. I see others are getting in the region of 100 calls per second, so am guessing my config has something suspect? And not just a 'tuning' issue as the difference is huge. CPU usage is <3% and hard disk sounds very active (however a 10ms operation every 200ms would sound busy to a human). So puzzles me why it is taking so long. My scenario is the SIPP provided UAC pointed to extension 3000 (the default set up conference with FS). Want to create a 100+ participant conference where 3 calls per second will not suffice. ;-( thanks, Niall. System Info: Installed from default Windows MSI installer. 6GB Ram Windows 7 64 Bit Intel Core i7 920 CPU From rupa at rupa.com Tue May 19 12:33:56 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 19 May 2009 14:33:56 -0500 Subject: [Freeswitch-users] Load test issue In-Reply-To: <4aec92830905191221m4e734f39wa3c2ef395bb99310@mail.gmail.com> References: <4aec92830905191221m4e734f39wa3c2ef395bb99310@mail.gmail.com> Message-ID: You probably wanna rip out all the PBX "features" and make it a dedicated conference system. On Tue, May 19, 2009 at 2:21 PM, Niall Crosby wrote: > Hi, > > Am running SIPP on a pretty fresh default install of FS and am only > getting 3 calls processed per second when put under load. > > I see others are getting in the region of 100 calls per second, so am > guessing my config has something suspect? And not just a 'tuning' > issue as the difference is huge. > > CPU usage is <3% and hard disk sounds very active (however a 10ms > operation every 200ms would sound busy to a human). So puzzles me why > it is taking so long. > > My scenario is the SIPP provided UAC pointed to extension 3000 (the > default set up conference with FS). > > Want to create a 100+ participant conference where 3 calls per second > will not suffice. ;-( > > thanks, > Niall. > > System Info: > Installed from default Windows MSI installer. > 6GB Ram > Windows 7 64 Bit > Intel Core i7 920 CPU > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/dfc10de7/attachment.html From brian at freeswitch.org Tue May 19 12:37:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 14:37:42 -0500 Subject: [Freeswitch-users] Load test issue In-Reply-To: References: <4aec92830905191221m4e734f39wa3c2ef395bb99310@mail.gmail.com> Message-ID: <845DE2A8-7F27-41EC-9EA7-1D1239EE4DA8@freeswitch.org> Or not load test against a conference... Its not a real test! /b On May 19, 2009, at 2:33 PM, Rupa Schomaker wrote: > You probably wanna rip out all the PBX "features" and make it a > dedicated conference system. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/54f26d7f/attachment.html From msc at freeswitch.org Tue May 19 12:48:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 May 2009 12:48:18 -0700 Subject: [Freeswitch-users] New ZDNet Article About FreeSWITCH Message-ID: <87f2f3b90905191248u2813e829veb4e7455b7de535e@mail.gmail.com> Gang, Please visit the main FreeSWITCH site and check out the linksto the new ZDNet story. Please spread the word with all of the link sharing sites that you have. Thanks! Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/7625ef69/attachment.html From brian at freeswitch.org Tue May 19 12:53:04 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 14:53:04 -0500 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> Message-ID: FreeSWITCHers, Is it too much to ask for the community to help promote the project when possible? Now I normally don't do this but I could put on a cheer leader outfit and grab some pomp pomps and get you guys and gals to click .... well maybe if you click I won't put on the outfit! :P Not sure which is more beneficial to you! Thanks, Brian West On May 19, 2009, at 12:28 PM, Anthony Minessale wrote: > please find it in your hearts to digg this story. > > http://digg.com/search?s=skype+switch > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/40d62392/attachment.html From erez at simonim.com Tue May 19 13:23:10 2009 From: erez at simonim.com (Erez Simon) Date: Tue, 19 May 2009 23:23:10 +0300 Subject: [Freeswitch-users] Freeswitch hosting Message-ID: <6a6efd0f0905191323x75373f81ve125b4bbb72be3f5@mail.gmail.com> Hi, I want to get a Freeswitch instance running on a hosted service, such as the Amazon EC2 cloud. Can anyone help in recommending Freeswitch hosting alternatives in terms of price, scale, support, quality, configurability, etc.? much obliged, Erez -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/ebbfbf36/attachment.html From sprice at gmail.com Tue May 19 13:38:18 2009 From: sprice at gmail.com (SP) Date: Tue, 19 May 2009 15:38:18 -0500 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> Message-ID: <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> I think we'd like to see the outfit! On Tue, May 19, 2009 at 14:53, Brian West wrote: > FreeSWITCHers, > Is it too much to ask for the community to help promote the project when > possible? ?Now I normally don't do this but I could put on a cheer leader > outfit and grab some pomp pomps and get you guys and gals to click .... well > maybe if you click I won't put on the outfit! ?:P ?Not sure which is more > beneficial to you! > Thanks, > Brian West > On May 19, 2009, at 12:28 PM, Anthony Minessale wrote: > > please find it in your hearts to digg this story. > > http://digg.com/search?s=skype+switch > > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Shannon From intralanman at freeswitch.org Tue May 19 13:44:24 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 19 May 2009 16:44:24 -0400 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> Message-ID: <4A131A28.90304@freeswitch.org> speak for yourself... the Dr. Pepper PJ's are bad enough ;-) -Ray SP wrote: > I think we'd like to see the outfit! > > On Tue, May 19, 2009 at 14:53, Brian West wrote: > >> FreeSWITCHers, >> Is it too much to ask for the community to help promote the project when >> possible? Now I normally don't do this but I could put on a cheer leader >> outfit and grab some pomp pomps and get you guys and gals to click .... well >> maybe if you click I won't put on the outfit! :P Not sure which is more >> beneficial to you! >> Thanks, >> Brian West >> On May 19, 2009, at 12:28 PM, Anthony Minessale wrote: >> >> please find it in your hearts to digg this story. >> >> http://digg.com/search?s=skype+switch >> >> >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/d267ea1b/attachment-0001.html From brian at freeswitch.org Tue May 19 13:44:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 15:44:02 -0500 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <4A131A28.90304@freeswitch.org> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> <4A131A28.90304@freeswitch.org> Message-ID: <2966A096-9661-40E0-A9D9-0F502D8ED29B@freeswitch.org> HAHAHAHAHAHAHAHAHAH /b On May 19, 2009, at 3:44 PM, Raymond Chandler wrote: > speak for yourself... the Dr. Pepper PJ's are bad enough ;-) > > -Ray Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/202559a8/attachment.html From niall.crosby at gmail.com Tue May 19 13:51:06 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Tue, 19 May 2009 21:51:06 +0100 Subject: [Freeswitch-users] Load test issue In-Reply-To: <845DE2A8-7F27-41EC-9EA7-1D1239EE4DA8@freeswitch.org> References: <4aec92830905191221m4e734f39wa3c2ef395bb99310@mail.gmail.com> <845DE2A8-7F27-41EC-9EA7-1D1239EE4DA8@freeswitch.org> Message-ID: <4aec92830905191351r28c2ff3ald73a10639243a5fa@mail.gmail.com> Thanks for the advice gents. Much appreciated. Removing all from the default dialplan with the exception of the conference ups the CPS to 20, much better. I also tried testing against the IVR demo (extension 5000) but no change, so appears not to be conference related. Although I'm happy with the current performance, is it normal my cpu is still showing < 4% usage through the test although FS appears to be hitting a limit of some sort? Am thinking it must be hard disk bottleneck but surely HD writes are quick??? 2009/5/19 Brian West : > Or not load test against a conference... Its not a real test! > /b > On May 19, 2009, at 2:33 PM, Rupa Schomaker wrote: > > You probably wanna rip out all the PBX "features" and make it a dedicated > conference system. > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- Sremium Ltd. Reg Number: 451937 Mobile: +353 (0)87 2393174 Web: www.sremium.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of Sremium. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. From brian at freeswitch.org Tue May 19 13:55:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 15:55:44 -0500 Subject: [Freeswitch-users] Load test issue In-Reply-To: <4aec92830905191351r28c2ff3ald73a10639243a5fa@mail.gmail.com> References: <4aec92830905191221m4e734f39wa3c2ef395bb99310@mail.gmail.com> <845DE2A8-7F27-41EC-9EA7-1D1239EE4DA8@freeswitch.org> <4aec92830905191351r28c2ff3ald73a10639243a5fa@mail.gmail.com> Message-ID: <14B5054C-9299-4C96-94ED-F10268F5AC9C@freeswitch.org> Try just playing a file... conference is heavy for load testing needs at those rates! /b On May 19, 2009, at 3:51 PM, Niall Crosby wrote: > Although I'm happy with the current performance, is it normal my cpu > is still showing < 4% usage through the test although FS appears to be > hitting a limit of some sort? Am thinking it must be hard disk > bottleneck but surely HD writes are quick??? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/f8fca121/attachment.html From msc at freeswitch.org Tue May 19 14:08:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 May 2009 14:08:57 -0700 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <2966A096-9661-40E0-A9D9-0F502D8ED29B@freeswitch.org> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> <4A131A28.90304@freeswitch.org> <2966A096-9661-40E0-A9D9-0F502D8ED29B@freeswitch.org> Message-ID: <87f2f3b90905191408v5e320275w369820fea18596e3@mail.gmail.com> i was kinda impressed with the dr. pepper pj's. :) On Tue, May 19, 2009 at 1:44 PM, Brian West wrote: > HAHAHAHAHAHAHAHAHAH > /b > > On May 19, 2009, at 3:44 PM, Raymond Chandler wrote: > > speak for yourself... the Dr. Pepper PJ's are bad enough ;-) > > -Ray > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/d0825028/attachment.html From msc at freeswitch.org Tue May 19 14:34:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 19 May 2009 14:34:01 -0700 Subject: [Freeswitch-users] Freeswitch hosting In-Reply-To: <6a6efd0f0905191323x75373f81ve125b4bbb72be3f5@mail.gmail.com> References: <6a6efd0f0905191323x75373f81ve125b4bbb72be3f5@mail.gmail.com> Message-ID: <87f2f3b90905191434m7f11826bk9c6c543f76c0b1a7@mail.gmail.com> Had you see this wiki page yet? http://wiki.freeswitch.org/wiki/Amazon_ec2 -MC On Tue, May 19, 2009 at 1:23 PM, Erez Simon wrote: > Hi, > > I want to get a Freeswitch instance running on a hosted service, such as > the Amazon EC2 cloud. > Can anyone help in recommending Freeswitch hosting alternatives in terms of > price, scale, support, quality, configurability, etc.? > > much obliged, > Erez > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/62b995bc/attachment.html From brian at freeswitch.org Tue May 19 14:37:24 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 19 May 2009 16:37:24 -0500 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <87f2f3b90905191408v5e320275w369820fea18596e3@mail.gmail.com> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> <4A131A28.90304@freeswitch.org> <2966A096-9661-40E0-A9D9-0F502D8ED29B@freeswitch.org> <87f2f3b90905191408v5e320275w369820fea18596e3@mail.gmail.com> Message-ID: <58C94E90-4A62-49CD-BCB2-0C8FA36F10AC@freeswitch.org> Well you can rest now... they were damaged in a washing machine accident! :( /b On May 19, 2009, at 4:08 PM, Michael Collins wrote: > i was kinda impressed with the dr. pepper pj's. :) > > On Tue, May 19, 2009 at 1:44 PM, Brian West > wrote: > HAHAHAHAHAHAHAHAHAH > > /b > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/e18b4a56/attachment-0001.html From anthony.minessale at gmail.com Tue May 19 14:43:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 19 May 2009 16:43:29 -0500 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <58C94E90-4A62-49CD-BCB2-0C8FA36F10AC@freeswitch.org> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> <4A131A28.90304@freeswitch.org> <2966A096-9661-40E0-A9D9-0F502D8ED29B@freeswitch.org> <87f2f3b90905191408v5e320275w369820fea18596e3@mail.gmail.com> <58C94E90-4A62-49CD-BCB2-0C8FA36F10AC@freeswitch.org> Message-ID: <191c3a030905191443l5bcc8e23mdffd54f0a4ed4407@mail.gmail.com> once we reach 1000 diggs, lets take up a collection to get him new ones! On Tue, May 19, 2009 at 4:37 PM, Brian West wrote: > Well you can rest now... they were damaged in a washing machine accident! > :( > /b > > On May 19, 2009, at 4:08 PM, Michael Collins wrote: > > i was kinda impressed with the dr. pepper pj's. :) > > On Tue, May 19, 2009 at 1:44 PM, Brian West wrote: > >> HAHAHAHAHAHAHAHAHAH >> /b >> >> > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/da360f2b/attachment.html From sprice at gmail.com Tue May 19 14:43:48 2009 From: sprice at gmail.com (SP) Date: Tue, 19 May 2009 16:43:48 -0500 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: <58C94E90-4A62-49CD-BCB2-0C8FA36F10AC@freeswitch.org> References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> <7e2ac3270905191338p7f9a7a6dm1480f16ff47ca43b@mail.gmail.com> <4A131A28.90304@freeswitch.org> <2966A096-9661-40E0-A9D9-0F502D8ED29B@freeswitch.org> <87f2f3b90905191408v5e320275w369820fea18596e3@mail.gmail.com> <58C94E90-4A62-49CD-BCB2-0C8FA36F10AC@freeswitch.org> Message-ID: <7e2ac3270905191443v1c8dd7aas8e4468b8761b98b2@mail.gmail.com> really?? is that your story you're sticking to?? On Tue, May 19, 2009 at 16:37, Brian West wrote: > Well you can rest now... they were damaged in a washing machine accident! > ?:( > /b > On May 19, 2009, at 4:08 PM, Michael Collins wrote: > > i was kinda impressed with the dr. pepper pj's. :) > > On Tue, May 19, 2009 at 1:44 PM, Brian West??wrote: >> >> HAHAHAHAHAHAHAHAHAH >> /b > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > -- Shannon From diego.viola at gmail.com Tue May 19 15:19:44 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 19 May 2009 18:19:44 -0400 Subject: [Freeswitch-users] As seen on DIGG In-Reply-To: References: <191c3a030905191028g3d289261q82d1202266abf6c6@mail.gmail.com> Message-ID: <86a32abc0905191519n7538d0bew4ba30e8cad7c0d59@mail.gmail.com> Digged. On Tue, May 19, 2009 at 3:53 PM, Brian West wrote: > FreeSWITCHers, > Is it too much to ask for the community to help promote the project when > possible? ?Now I normally don't do this but I could put on a cheer leader > outfit and grab some pomp pomps and get you guys and gals to click .... well > maybe if you click I won't put on the outfit! ?:P ?Not sure which is more > beneficial to you! > Thanks, > Brian West > On May 19, 2009, at 12:28 PM, Anthony Minessale wrote: > > please find it in your hearts to digg this story. > > http://digg.com/search?s=skype+switch > > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Tue May 19 16:15:32 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 19 May 2009 19:15:32 -0400 Subject: [Freeswitch-users] Freeswitch hosting In-Reply-To: <87f2f3b90905191434m7f11826bk9c6c543f76c0b1a7@mail.gmail.com> References: <6a6efd0f0905191323x75373f81ve125b4bbb72be3f5@mail.gmail.com> <87f2f3b90905191434m7f11826bk9c6c543f76c0b1a7@mail.gmail.com> Message-ID: <86a32abc0905191615l4c059764od92d9552f2a3ed6c@mail.gmail.com> FreeSWITCH works nice on Amazon EC2, you just have to open the SIP and RTP ports and set ext-sip-ip and ext-rtp-ip to the external EC2 ip's in your Sofia profile. Diego On Tue, May 19, 2009 at 5:34 PM, Michael Collins wrote: > Had you see this wiki page yet? > http://wiki.freeswitch.org/wiki/Amazon_ec2 > > -MC > > On Tue, May 19, 2009 at 1:23 PM, Erez Simon wrote: >> >> Hi, >> >> I want to?get a?Freeswitch instance running on a hosted service, such as >> the Amazon EC2 cloud. >> Can?anyone help in recommending?Freeswitch hosting alternatives in terms >> of price, scale, support, quality, configurability, etc.? >> >> much obliged, >> Erez >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Tue May 19 16:18:42 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 19 May 2009 19:18:42 -0400 Subject: [Freeswitch-users] Testimonials In-Reply-To: <23611131.post@talk.nabble.com> References: <23611131.post@talk.nabble.com> Message-ID: <86a32abc0905191618t3821f60cv7c2e62eb4ef7fb41@mail.gmail.com> Hi Maxim, We are using FreeSWITCH as an office PBX in Rich Apps Consulting (http://www.richappsconsulting.com/). I have written a blog post entry a long time ago, it's pretty old by now, but I guess still relevant. http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ It works great, I also have commented on the testimonial article, in the wiki about it. http://wiki.freeswitch.org/wiki/Testimonials#Office_PBX Regards, Diego On Tue, May 19, 2009 at 3:49 AM, Maxim Tsvetov wrote: > > Hello, > > Our company want to use Freeswitch and now we testing this solution in the > lab. > > If someone already using Freeswitch as office pbx, ivr or any other > commercial purposes > could you please share you experience (and if it is not a secret - company > name)? > > Regards, > Maxim Tsvetov > -- > View this message in context: http://www.nabble.com/Testimonials-tp23611131p23611131.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Tue May 19 16:19:18 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 19 May 2009 19:19:18 -0400 Subject: [Freeswitch-users] Testimonials In-Reply-To: <86a32abc0905191618t3821f60cv7c2e62eb4ef7fb41@mail.gmail.com> References: <23611131.post@talk.nabble.com> <86a32abc0905191618t3821f60cv7c2e62eb4ef7fb41@mail.gmail.com> Message-ID: <86a32abc0905191619t638adcaat7eb617790ea2bae5@mail.gmail.com> I'm also looking to install more PBXs with it soon, because FreeSWITCH does a great job at that, and at everything else too :) Diego On Tue, May 19, 2009 at 7:18 PM, Diego Viola wrote: > Hi Maxim, > > We are using FreeSWITCH as an office PBX in Rich Apps Consulting > (http://www.richappsconsulting.com/). > > I have written a blog post entry a long time ago, it's pretty old by > now, but I guess still relevant. > http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ > > It works great, I also have commented on the testimonial article, in > the wiki about it. > http://wiki.freeswitch.org/wiki/Testimonials#Office_PBX > > Regards, > > Diego > > On Tue, May 19, 2009 at 3:49 AM, Maxim Tsvetov wrote: >> >> Hello, >> >> Our company want to use Freeswitch and now we testing this solution in the >> lab. >> >> If someone already using Freeswitch as office pbx, ivr or any other >> commercial purposes >> could you please share you experience (and if it is not a secret - company >> name)? >> >> Regards, >> Maxim Tsvetov >> -- >> View this message in context: http://www.nabble.com/Testimonials-tp23611131p23611131.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From red.rain.seven at gmail.com Tue May 19 17:39:50 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Tue, 19 May 2009 17:39:50 -0700 Subject: [Freeswitch-users] Testimonials In-Reply-To: <86a32abc0905191619t638adcaat7eb617790ea2bae5@mail.gmail.com> References: <23611131.post@talk.nabble.com> <86a32abc0905191618t3821f60cv7c2e62eb4ef7fb41@mail.gmail.com> <86a32abc0905191619t638adcaat7eb617790ea2bae5@mail.gmail.com> Message-ID: <59ad9ca10905191739n4e57eedk54682c80428e7d2e@mail.gmail.com> Deigo: Do you work on some GUI interface or purely text environment? On Tue, May 19, 2009 at 4:19 PM, Diego Viola wrote: > I'm also looking to install more PBXs with it soon, because FreeSWITCH > does a great job at that, and at everything else too :) > > Diego > > On Tue, May 19, 2009 at 7:18 PM, Diego Viola > wrote: > > Hi Maxim, > > > > We are using FreeSWITCH as an office PBX in Rich Apps Consulting > > (http://www.richappsconsulting.com/). > > > > I have written a blog post entry a long time ago, it's pretty old by > > now, but I guess still relevant. > > > http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ > > > > It works great, I also have commented on the testimonial article, in > > the wiki about it. > > http://wiki.freeswitch.org/wiki/Testimonials#Office_PBX > > > > Regards, > > > > Diego > > > > On Tue, May 19, 2009 at 3:49 AM, Maxim Tsvetov > wrote: > >> > >> Hello, > >> > >> Our company want to use Freeswitch and now we testing this solution in > the > >> lab. > >> > >> If someone already using Freeswitch as office pbx, ivr or any other > >> commercial purposes > >> could you please share you experience (and if it is not a secret - > company > >> name)? > >> > >> Regards, > >> Maxim Tsvetov > >> -- > >> View this message in context: > http://www.nabble.com/Testimonials-tp23611131p23611131.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/602b80e5/attachment-0001.html From edpimentl at gmail.com Tue May 19 17:53:14 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 19 May 2009 20:53:14 -0400 Subject: [Freeswitch-users] Testimonials In-Reply-To: <1242754173.20213.5.camel@dk-d820> References: <23611131.post@talk.nabble.com> <1242754173.20213.5.camel@dk-d820> Message-ID: <9dc4a1670905191753m5fa66773v49b63f49a4cce02c@mail.gmail.com> Could not registered, without a invite code. Do u have one? Best regards, -E http://AskTwitR.com (Real Time Twitter Search & Reputation Management) http://TwiTR.Me (Cross Social Network Messaging Bus) http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://TweetUp.ws (Twitter based MeetUp service) http://PiCurio.us (Spell with FlickR, Let others SEE what you are saying) http://TweetOrder.in/ (Tweet Food Order) http://DatR.Ws (Cloud Computing Media Sharing, Access and Publish) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/203753ec/attachment.html From diego.viola at gmail.com Tue May 19 18:10:03 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 19 May 2009 21:10:03 -0400 Subject: [Freeswitch-users] Testimonials In-Reply-To: <59ad9ca10905191739n4e57eedk54682c80428e7d2e@mail.gmail.com> References: <23611131.post@talk.nabble.com> <86a32abc0905191618t3821f60cv7c2e62eb4ef7fb41@mail.gmail.com> <86a32abc0905191619t638adcaat7eb617790ea2bae5@mail.gmail.com> <59ad9ca10905191739n4e57eedk54682c80428e7d2e@mail.gmail.com> Message-ID: <86a32abc0905191810y100a3fc6if37f4555aed74586@mail.gmail.com> I work with the standard CLI tools (command line interface). Best, Diego On Tue, May 19, 2009 at 8:39 PM, Henry Huang wrote: > Deigo: > > Do you work on some GUI interface or purely text environment? > > On Tue, May 19, 2009 at 4:19 PM, Diego Viola wrote: >> >> I'm also looking to install more PBXs with it soon, because FreeSWITCH >> does a great job at that, and at everything else too :) >> >> Diego >> >> On Tue, May 19, 2009 at 7:18 PM, Diego Viola >> wrote: >> > Hi Maxim, >> > >> > We are using FreeSWITCH as an office PBX in Rich Apps Consulting >> > (http://www.richappsconsulting.com/). >> > >> > I have written a blog post entry a long time ago, it's pretty old by >> > now, but I guess still relevant. >> > >> > http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ >> > >> > It works great, I also have commented on the testimonial article, in >> > the wiki about it. >> > http://wiki.freeswitch.org/wiki/Testimonials#Office_PBX >> > >> > Regards, >> > >> > Diego >> > >> > On Tue, May 19, 2009 at 3:49 AM, Maxim Tsvetov >> > wrote: >> >> >> >> Hello, >> >> >> >> Our company want to use Freeswitch and now we testing this solution in >> >> the >> >> lab. >> >> >> >> If someone already using Freeswitch as office pbx, ivr or any other >> >> commercial purposes >> >> could you please share you experience (and if it is not a secret - >> >> company >> >> name)? >> >> >> >> Regards, >> >> Maxim Tsvetov >> >> -- >> >> View this message in context: >> >> http://www.nabble.com/Testimonials-tp23611131p23611131.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ronmccar at gmail.com Tue May 19 18:18:01 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Tue, 19 May 2009 18:18:01 -0700 Subject: [Freeswitch-users] Logging 503's or other errors In-Reply-To: References: <3885f4fe0905171126y7eb9b419x4c103a53cf4a6cad@mail.gmail.com> <191c3a030905180605w31b6114aoceabb866108b2a7@mail.gmail.com> Message-ID: <3885f4fe0905191818h73dfd2fctba0805510c64cee2@mail.gmail.com> Intresting, I will have to look at both of these. We route advance based on a 503 so we don't even a pipe we just go down in order. Just had a ideal, could you run some scripts between the next gateway, as in the next line in the XML calls a script or something that would then log it? Thanks On Mon, May 18, 2009 at 12:21 PM, dujinfang wrote: > Even the b leg cdr is enabled it only remember the final state(channel > vars) on the b leg. > > At least there are two possible ways to keep tracking all the gateways: > > 1) don't use '|' separated dial string, use a lua script like this: > > session:execute("bridge", dial_string1); > bridge_hangup_cause = session:getVariable("bridge_hangup_cause") or > session:getVariable("originate_disposition"); > if (bridge_hangup_cause == "NORMAL_TEMPORARY_FAILURE" or > bridge_hangup_cause == "NO_ROUTE_DESTINATION" or bridge_hangup_cause == > "CALL_REJECTED") then > freeswitch.consoleLog("notice", "Hangup. Cause: [" .. > bridge_hangup_cause .. "]. Retry: " > -- database.insert('something') > > session:execue("bridge", dial_string2); > if (bridge_hangup_cause == "NORMAL_TEMPORARY_FAILURE" or > bridge_hangup_cause == "NO_ROUTE_DESTINATION" or bridge_hangup_cause == > "CALL_REJECTED") then > session:execute("bridge", dial_string3); > .... obviously it can be done in a loop > > 2) by sip: add a custom header to INVITE, > > bridge({sip_h_x_xxx=yyy}sofia/gateways/a/...|sofia/gateways/b/...|sofia/gateways/c/... > > be sure to give yyy a unique value each time you call, then you can dump > all the sip messages and by cross reference of the sip_h_x_xxx and call-ID > you can get all the related sip messages(every INVITE will have the same > sip_h_x_xxx header and each INVITE related message will have the same > call-ID. > > > > > On May 18, 2009, at 9:05 PM, Anthony Minessale wrote: > > enable the b leg cdr as well and you will also get cdr from the b leg > perspective. > both xml cdr and cdr csv have params in the config to enable it. > > > On Sun, May 17, 2009 at 1:26 PM, Ron McCarthy wrote: > >> Hi list, >> >> Ive been trying to find a way to log 503's, 480's and other SIP response >> codes. If we have continue_on_fail=true and have multiple gateways for the >> call to go out, if the 1st,2nd or whatever gateways fail can we log it >> somehow? We'd like to know if a carrier is having issues or not letting us >> send calls for some reason, from what I can tell I only show one CDR get >> written and that's at the end of the call, so it says nothing about the >> gateways we tried to send a call before and if they failed. >> >> Any ideals on how to do this? Im using the XML CURL dialplan if that >> matter. Any ideals how this could be setup so we can keep track of what is >> going on? >> >> Thanks >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090519/73af234d/attachment.html From jim at evolutiontel.net Tue May 19 21:45:00 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 20 May 2009 14:45:00 +1000 Subject: [Freeswitch-users] Dialplan action sched_hangup Message-ID: Hi All, Can someone clarify the usage of the 'sched_hangup' application in mod_dptools. In the wiki it suggests the timer starts when the call is answered, however testing proves this not to be the case and the call is released while in the ringing state if the timer is short enough. Or should it be used in conjuction with channel variable 'execute_on_answer'. Thus forcing setting of the timer at this point of the call/ If you need the debug or any other info please let me know. Thanks, Jim From jim at evolutiontel.net Tue May 19 22:48:25 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 20 May 2009 15:48:25 +1000 Subject: [Freeswitch-users] Unable to successfully bridge calls to an "external" user In-Reply-To: <27414EDF-1B3C-4453-962F-0C6CAD1D46F3@freeswitch.org> References: <651FE195-03AC-4D51-957B-B6EF463939AB@gmail.com> <191c3a030905190556l8ffd561h19bafa90a5369b9c@mail.gmail.com> <27414EDF-1B3C-4453-962F-0C6CAD1D46F3@freeswitch.org> Message-ID: Hey David, IMHO first you need to decide if you want to proxy the media traffic or not (look at bypass_media), as you are behind a NAT it suggests that you are perhaps using a cable or adsl connection to the internet and may not want to give up some of your bandwidth for VOIP calls to external connections. If you choose to bypass the media, you will then need to make sure the IP address reported in the 200 OK by the terminating user on answer is reported correctly to Faktortel your ITSP. You might find this mode will work as Faktortel will probably be able to determine the path to the terminating phone based on the IP and PORT it received the voice packets from. Alternatively if you want to proxy the media traffic, you will need to make sure that FS reports the correct External IP address in the INVITE message to the terminating user. These settings are mentioned by Anthony below. I use both NGREP and TCPDUMP heavily when trying new things on FS, because when you determine what comes out gets easier to findout what parms to change. Regards, Jim On Tue, May 19, 2009 at 11:24 PM, Brian West wrote: > You will also need to modify the dial-string in conf/directory/default.xml > because it only looks on internal for registered users. > /b > On May 19, 2009, at 7:56 AM, Anthony Minessale wrote: > > edit your sip profile and comment out every line that contains the string > nat to disable all the nat auto-detection. > for dmz, you need to set the rtp-ext-ip and sip-ext-ip to be the live ip and > sip-ip and rtp-ip to be the lan ip (the real one) > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dujinfang at gmail.com Tue May 19 22:51:57 2009 From: dujinfang at gmail.com (seven) Date: Wed, 20 May 2009 13:51:57 +0800 Subject: [Freeswitch-users] Dialplan action sched_hangup In-Reply-To: References: Message-ID: <5825B5E5-91B1-4ADF-B3A0-1F20A5B6EE4D@gmail.com> try ignore_early_media On May 20, 2009, at 12:45 PM, Jim Burke wrote: > Hi All, > > Can someone clarify the usage of the 'sched_hangup' application in > mod_dptools. In the wiki it suggests the timer starts when the call > is answered, however testing proves this not to be the case and the > call is released while in the ringing state if the timer is short > enough. > > Or should it be used in conjuction with channel variable > 'execute_on_answer'. Thus forcing setting of the timer at this point > of the call/ > > If you need the debug or any other info please let me know. > > Thanks, > Jim > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jim at evolutiontel.net Tue May 19 23:15:49 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 20 May 2009 16:15:49 +1000 Subject: [Freeswitch-users] Dialplan action sched_hangup In-Reply-To: <5825B5E5-91B1-4ADF-B3A0-1F20A5B6EE4D@gmail.com> References: <5825B5E5-91B1-4ADF-B3A0-1F20A5B6EE4D@gmail.com> Message-ID: Hi Seven, Thanks, however there is no early_media in this call. xlite----FS---eyebeam. bypass_media is set and the response to the invite is 180 ringing. Regards, Jim On Wed, May 20, 2009 at 3:51 PM, seven wrote: > try ignore_early_media > > On May 20, 2009, at 12:45 PM, Jim Burke wrote: > >> Hi All, >> >> Can someone clarify the usage of the 'sched_hangup' application in >> mod_dptools. ?In the wiki it suggests the timer starts when the call >> is answered, however testing proves this not to be the case and the >> call is released while in the ringing state if the timer is short >> enough. >> >> Or should it be used in conjuction with channel variable >> 'execute_on_answer'. ?Thus forcing setting of the timer at this point >> of the call/ >> >> If you need the debug or any other info please let me know. >> >> Thanks, >> Jim >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ravi_hum at yahoo.co.in Tue May 19 23:12:54 2009 From: ravi_hum at yahoo.co.in (ravi hum) Date: Tue, 19 May 2009 23:12:54 -0700 (PDT) Subject: [Freeswitch-users] unable to recieve audio on endpoints Message-ID: <23629268.post@talk.nabble.com> I have installed Freeswitch and configured to use SIP Proxy Server for making calls. I tested Originate and Bridge commands and both sides of the call are getting connected. Now the problem is :- --I am not recieving any voice/audio on either endpoint of the call ( both endpoints are mobile number) --firewall is off on the server *call request is send from FreeSwitch to SIP Proxy server ( FreeSwitch --->> SIP Proxy Server) please let me know how to solve this issue. -- View this message in context: http://www.nabble.com/unable-to-recieve-audio-on-endpoints-tp23629268p23629268.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Wed May 20 01:10:11 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 20 May 2009 18:10:11 +1000 Subject: [Freeswitch-users] unable to recieve audio on endpoints In-Reply-To: <23629268.post@talk.nabble.com> References: <23629268.post@talk.nabble.com> Message-ID: <20090520081011.GA11923@jdc.jasonjgw.net> ravi hum wrote: > *call request is send from FreeSwitch to SIP Proxy server ( FreeSwitch --->> > SIP Proxy Server) > please let me know how to solve this issue. If there is a NAT device involved anywhere in your scenario, it's probably the cause. http://wiki.freeswitch.org/wiki/NAT RTP would appear to be the issue in your case. If there is no NAT anywhere, then it's time to do some network debugging with your favourite packet monitoring tool. From fdelawarde at wirelessmundi.com Wed May 20 03:26:18 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 20 May 2009 12:26:18 +0200 Subject: [Freeswitch-users] openzap and progress detection In-Reply-To: <1242635370.17063.67.camel@localhost.localdomain> References: <1242635370.17063.67.camel@localhost.localdomain> Message-ID: <1242815178.3030.10.camel@localhost.localdomain> I briefly looked at the code, so I will then answer my own questions from what i understood (and I didn't understand much). Please correct me if I'm wrong (see inline answers). A new question would then be: Are any of those features planned for the future of OpenZAP? On Mon, 2009-05-18 at 10:29 +0200, Francois Delawarde wrote: > Hello, > > I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO > +1xFXS), and am trying first to make the FXO work with Openzap and > Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and > loads the spans, but I'm currently enable to dial out with the FXO > module, it doesn't dial anything and times-out after 30 seconds. I > believe it has to do with some tone detection and therefore have a > few questions: > > - When I plug in the line, dahdi sends an event (event 17) that is > ignored by OZ. Can we enable some type of battery check in OZ before > dialing out, or is there some variable to monitor battery (oz dump > doesn't show the battery status)? No. > - Does OZ use the polarity switch events sent by dahdi (in kewlstart > mode) for answer and hangup detection? Yes. > - Apparently, OZ does tone progress detection by frequency, but here > in Spain, most tones use the same 425Hz frequency with different > on-off timing. Is it possible to detect those? No. > - As some PBX in Spain transfer calls by first hanging up and picking > up on another phone, can we enable/disable parts of polarity switch > and/or tone progress detection (ex: (hangup)/(answer)onpolarityswitch > in Asterisk) No. > - Some lines here are connected to very old FXS from operators that > have low sound quality and can take a few seconds to give a dial tone > when picking up. Is it possible to introduce a delay before sending > DTMF digits when dialing? Is it possible to "relax" DTMF detection, > and tweak DTMF settings (make them a bit longer, with a longer pause > for the other side to detect)? No / No idea / No idea > My "ideal" case to make it work in every case around here would be to: > - have OZ fail to dial if battery is not present (and be able to fetch > battery status somehow) > - disable tone progress (sometimes call ends up on some local PBX that > answers and provides US tones which are different) > - be able to have an initial pause before dialing with DTMF digits > - use polarity switch to detect remote answer, but not hangup (for > transfer issues) > > Is the above possible? No. > Thanks in advance, > Fran?ois. You're welcome! Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/242f894f/attachment.html From anthony.minessale at gmail.com Wed May 20 05:49:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 20 May 2009 07:49:55 -0500 Subject: [Freeswitch-users] Dialplan action sched_hangup In-Reply-To: References: <5825B5E5-91B1-4ADF-B3A0-1F20A5B6EE4D@gmail.com> Message-ID: <191c3a030905200549m6d6f9292v19d44e8ea81e509@mail.gmail.com> the timer starts as soon as you make the call to sched_hangup if you want to start it as soon as the call is answered run the command via the execute_on_answer mechanism On Wed, May 20, 2009 at 1:15 AM, Jim Burke wrote: > Hi Seven, > > Thanks, however there is no early_media in this call. > xlite----FS---eyebeam. bypass_media is set and the response to the > invite is 180 ringing. > > Regards, > Jim > > On Wed, May 20, 2009 at 3:51 PM, seven wrote: > > try ignore_early_media > > > > On May 20, 2009, at 12:45 PM, Jim Burke wrote: > > > >> Hi All, > >> > >> Can someone clarify the usage of the 'sched_hangup' application in > >> mod_dptools. In the wiki it suggests the timer starts when the call > >> is answered, however testing proves this not to be the case and the > >> call is released while in the ringing state if the timer is short > >> enough. > >> > >> Or should it be used in conjuction with channel variable > >> 'execute_on_answer'. Thus forcing setting of the timer at this point > >> of the call/ > >> > >> If you need the debug or any other info please let me know. > >> > >> Thanks, > >> Jim > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/bdcea298/attachment.html From anthony.minessale at gmail.com Wed May 20 06:08:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 20 May 2009 08:08:23 -0500 Subject: [Freeswitch-users] openzap and progress detection In-Reply-To: <1242815178.3030.10.camel@localhost.localdomain> References: <1242635370.17063.67.camel@localhost.localdomain> <1242815178.3030.10.camel@localhost.localdomain> Message-ID: <191c3a030905200608v8d8b68eq695863404604796a@mail.gmail.com> The most difficult one would be the cadence detection. The rest are just based on events from the driver. We have never tried it in Spain so we unfortunately have not had a working environment to test it in. I am sure we could strive to support your requests but it will take time and resources so maybe you can open a feature request on jira http://jira.freeswitch.organd we can try to keep track of your endeavor. On Wed, May 20, 2009 at 5:26 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > I briefly looked at the code, so I will then answer my own questions from > what i understood (and I didn't understand much). Please correct me if I'm > wrong (see inline answers). > > A new question would then be: Are any of those features planned for the > future of OpenZAP? > > > On Mon, 2009-05-18 at 10:29 +0200, Francois Delawarde wrote: > > Hello, > > I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO+1xFXS), > and am trying first to make the FXO work with Openzap and Freeswitch (using > dahdi 2.2.0-rc4). Openzap perfectly detects and loads the spans, but I'm > currently enable to dial out with the FXO module, it doesn't dial anything > and times-out after 30 seconds. I believe it has to do with some tone > detection and therefore have a few questions: > > - When I plug in the line, dahdi sends an event (event 17) that is ignored > by OZ. Can we enable some type of battery check in OZ before dialing out, or > is there some variable to monitor battery (oz dump doesn't show the battery > status)? > > No. > > - Does OZ use the polarity switch events sent by dahdi (in kewlstart mode) > for answer and hangup detection? > > Yes. > > - Apparently, OZ does tone progress detection by frequency, but here in > Spain, most tones use the same 425Hz frequency with different on-off timing. > Is it possible to detect those? > > No. > > - As some PBX in Spain transfer calls by first hanging up and picking up > on another phone, can we enable/disable parts of polarity switch and/or tone > progress detection (ex: (hangup)/(answer)onpolarityswitch in Asterisk) > > No. > > - Some lines here are connected to very old FXS from operators that have > low sound quality and can take a few seconds to give a dial tone when > picking up. Is it possible to introduce a delay before sending DTMF digits > when dialing? Is it possible to "relax" DTMF detection, and tweak DTMF > settings (make them a bit longer, with a longer pause for the other side to > detect)? > > No / No idea / No idea > > My "ideal" case to make it work in every case around here would be to: > - have OZ fail to dial if battery is not present (and be able to fetch > battery status somehow) > - disable tone progress (sometimes call ends up on some local PBX that > answers and provides US tones which are different) > - be able to have an initial pause before dialing with DTMF digits > - use polarity switch to detect remote answer, but not hangup (for transfer > issues) > > Is the above possible? > > No. > > Thanks in advance, > Fran?ois. > > You're welcome! > > Fran?ois. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/cb530374/attachment-0001.html From jim at evolutiontel.net Wed May 20 06:34:31 2009 From: jim at evolutiontel.net (Jim Burke) Date: Wed, 20 May 2009 23:34:31 +1000 Subject: [Freeswitch-users] Dialplan action sched_hangup In-Reply-To: <191c3a030905200549m6d6f9292v19d44e8ea81e509@mail.gmail.com> References: <5825B5E5-91B1-4ADF-B3A0-1F20A5B6EE4D@gmail.com> <191c3a030905200549m6d6f9292v19d44e8ea81e509@mail.gmail.com> Message-ID: Thanks Anthony, I will give it a go. Cheers, On Wed, May 20, 2009 at 10:49 PM, Anthony Minessale wrote: > the timer starts as soon as you make the call to sched_hangup > if you want to start it as soon as the call is answered run the command via > the execute_on_answer mechanism > > On Wed, May 20, 2009 at 1:15 AM, Jim Burke wrote: >> >> Hi Seven, >> >> Thanks, however there is no early_media in this call. >> xlite----FS---eyebeam. ?bypass_media is set and the response to the >> invite is 180 ringing. >> >> Regards, >> Jim >> >> On Wed, May 20, 2009 at 3:51 PM, seven wrote: >> > try ignore_early_media >> > >> > On May 20, 2009, at 12:45 PM, Jim Burke wrote: >> > >> >> Hi All, >> >> >> >> Can someone clarify the usage of the 'sched_hangup' application in >> >> mod_dptools. ?In the wiki it suggests the timer starts when the call >> >> is answered, however testing proves this not to be the case and the >> >> call is released while in the ringing state if the timer is short >> >> enough. >> >> >> >> Or should it be used in conjuction with channel variable >> >> 'execute_on_answer'. ?Thus forcing setting of the timer at this point >> >> of the call/ >> >> >> >> If you need the debug or any other info please let me know. >> >> >> >> Thanks, >> >> Jim >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mariusz_kolo at wp.pl Wed May 20 05:48:53 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk_WP?=) Date: Wed, 20 May 2009 14:48:53 +0200 Subject: [Freeswitch-users] Core Dump with signal 6, Aborted Message-ID: <4A13FC35.9030200@wp.pl> Hello I have a problem with core dumped with signal 6 Aborted in gdb looks: Core was generated by `/usr/local/freeswitch//bin/freeswitch'. Program terminated with signal 6, Aborted. .... .... .... #0 0xb7f4f410 in __kernel_vsyscall () #1 0xb7c11085 in raise () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7c12a01 in abort () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7c49b7c in ?? () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7cd3138 in __fortify_fail () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7cd30f0 in __stack_chk_fail () from /lib/tls/i686/cmov/libc.so.6 #6 0xb7ee6f64 in __stack_chk_fail_local () from /usr/local/freeswitch/lib/libfreeswitch.so.1 #7 0xb7e345d8 in record_callback (bug=0xb01c0488, user_data=0xa73892b8, type=63780) at src/switch_ivr_async.c:455 #8 0x0000f844 in ?? () Backtrace stopped: previous frame inner to this frame (corrupt stack?) FreeSwitch version: FreeSWITCH Version 1.0.3 (exported) Linux: Linux pbx 2.6.24-23-server #1 SMP Mon Jan 26 00:55:21 UTC 2009 i686 GNU/Linux I'm using ISDN PRI via digium card and sip gateways This problem occurs when i'm using only voip for outbound calls. By default inbound calls alway comes from ISDN PRI. Thanks From brian at freeswitch.org Wed May 20 07:30:29 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 20 May 2009 09:30:29 -0500 Subject: [Freeswitch-users] Core Dump with signal 6, Aborted In-Reply-To: <4A13FC35.9030200@wp.pl> References: <4A13FC35.9030200@wp.pl> Message-ID: I would update to SVN trunk and try again chances are this has already been fixed. If it happens after you update please report bugs to http://jira.freeswitch.org Thanks, Brian On May 20, 2009, at 7:48 AM, Mariusz Ko?odziejczyk WP wrote: > Hello > > I have a problem with core dumped with signal 6 Aborted > > in gdb looks: > > Core was generated by `/usr/local/freeswitch//bin/freeswitch'. > Program terminated with signal 6, Aborted. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/6e9536ad/attachment.html From fdelawarde at wirelessmundi.com Wed May 20 08:01:55 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 20 May 2009 17:01:55 +0200 Subject: [Freeswitch-users] openzap and progress detection In-Reply-To: <191c3a030905200608v8d8b68eq695863404604796a@mail.gmail.com> References: <1242635370.17063.67.camel@localhost.localdomain> <1242815178.3030.10.camel@localhost.localdomain> <191c3a030905200608v8d8b68eq695863404604796a@mail.gmail.com> Message-ID: <1242831715.3545.33.camel@localhost.localdomain> Ok, feature request for progress detection opened as OPENZAP-70. Meanwhile, I humbly offer a basic working environment in Spain (a TDM400 with 1xFXO) if you need it for testing. Fran?ois. On Wed, 2009-05-20 at 08:08 -0500, Anthony Minessale wrote: > The most difficult one would be the cadence detection. > The rest are just based on events from the driver. We have never > tried it in Spain so we unfortunately have not had > a working environment to test it in. I am sure we could strive to > support your requests but it will take time and resources > so maybe you can open a feature request on jira > http://jira.freeswitch.org and we can try to keep track of your > endeavor. > > > > > On Wed, May 20, 2009 at 5:26 AM, Fran?ois Delawarde > wrote: > > I briefly looked at the code, so I will then answer my own > questions from what i understood (and I didn't understand > much). Please correct me if I'm wrong (see inline answers). > > A new question would then be: Are any of those features > planned for the future of OpenZAP? > > > On Mon, 2009-05-18 at 10:29 +0200, Francois Delawarde wrote: > > > Hello, > > > > I'm in Spain with an analog TDM400 Clone from OpenVox (with > > 1xFXO+1xFXS), and am trying first to make the FXO work with > > Openzap and Freeswitch (using dahdi 2.2.0-rc4). Openzap > > perfectly detects and loads the spans, but I'm currently > > enable to dial out with the FXO module, it doesn't dial > > anything and times-out after 30 seconds. I believe it has > > to do with some tone detection and therefore have a few > > questions: > > > > - When I plug in the line, dahdi sends an event (event 17) > > that is ignored by OZ. Can we enable some type of battery > > check in OZ before dialing out, or is there some variable to > > monitor battery (oz dump doesn't show the battery status)? > > No. > > > > - Does OZ use the polarity switch events sent by dahdi (in > > kewlstart mode) for answer and hangup detection? > > Yes. > > > > - Apparently, OZ does tone progress detection by frequency, > > but here in Spain, most tones use the same 425Hz frequency > > with different on-off timing. Is it possible to detect > > those? > > No. > > > > - As some PBX in Spain transfer calls by first hanging up > > and picking up on another phone, can we enable/disable parts > > of polarity switch and/or tone progress detection (ex: > > (hangup)/(answer)onpolarityswitch in Asterisk) > > No. > > > > - Some lines here are connected to very old FXS from > > operators that have low sound quality and can take a few > > seconds to give a dial tone when picking up. Is it possible > > to introduce a delay before sending DTMF digits when > > dialing? Is it possible to "relax" DTMF detection, and tweak > > DTMF settings (make them a bit longer, with a longer pause > > for the other side to detect)? > > No / No idea / No idea > > > > My "ideal" case to make it work in every case around here > > would be to: > > - have OZ fail to dial if battery is not present (and be > > able to fetch battery status somehow) > > - disable tone progress (sometimes call ends up on some > > local PBX that answers and provides US tones which are > > different) > > - be able to have an initial pause before dialing with DTMF > > digits > > - use polarity switch to detect remote answer, but not > > hangup (for transfer issues) > > > > Is the above possible? > > No. > > > > Thanks in advance, > > Fran?ois. > > You're welcome! > > Fran?ois. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/5cde3a3c/attachment.html From yudha2008 at gmail.com Wed May 20 08:40:50 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 20 May 2009 21:10:50 +0530 Subject: [Freeswitch-users] JavaScript session Transfer Message-ID: *Hi, I have an issue in transfer the call through JavaScript session. In JavaScript Session i have dialed 2 numbers One is mobile number and another one is extension number I want both the call to transfer in the conference room using JavaScript session Whether transfer is possible in JavaScript session . If possible means how? Can any one assist me to solve above process. Thanks in advance. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/d9ca86b1/attachment-0001.html From chris at maxpowersoft.com Wed May 20 08:53:38 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 20 May 2009 08:53:38 -0700 Subject: [Freeswitch-users] JavaScript session Transfer In-Reply-To: References: Message-ID: <4A142782.6030404@maxpowersoft.com> Hi N.Baskar, Create an extension mapping in your dialplan for your conference room as follows: (make sure this is accessible for both your local extensions and your public numbers) Within your JavaScript code do something like this: var conferenceRoomID = "666"; var trans = "testConf"+conferenceRoomID; session.execute("transfer", trans); That should do it. Regards, Chris ** Baskar wrote: > *Hi, > > I have an issue in transfer the call through JavaScript session. > > In JavaScript Session i have dialed 2 numbers > > One is mobile number and another one is extension number > > I want both the call to transfer in the conference room using > JavaScript session > > Whether transfer is possible in JavaScript session . If possible means > how? > > Can any one assist me to solve above process. > > Thanks in advance. > > -- > Warm Regards, > N.Baskar > > * > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/1e140cf5/attachment.html From lcm at marshap.com Wed May 20 10:42:45 2009 From: lcm at marshap.com (Larry Marshall) Date: Wed, 20 May 2009 10:42:45 -0700 Subject: [Freeswitch-users] Retrieving voicemail Message-ID: <001801c9d972$6272ce40$27586ac0$@com> In the http://wiki.freeswitch.org/wiki/Getting_Started_Guide under Basic Calling, it states "Call your own extension number to login to your voicemail box". When I dial my extension (1000) from an internal softphone at the same extension, it just asks me to leave message. It does not give me the opportunity to hear my messages. What am I missing. I am using the default configuration. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/b390de59/attachment.html From msc at freeswitch.org Wed May 20 12:04:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 May 2009 12:04:03 -0700 Subject: [Freeswitch-users] Retrieving voicemail In-Reply-To: <001801c9d972$6272ce40$27586ac0$@com> References: <001801c9d972$6272ce40$27586ac0$@com> Message-ID: <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> Lars, Thanks for pointing this out. I will update the wiki. The new way to check voicemail is to dial 4000 and then enter your extension. -MC On Wed, May 20, 2009 at 10:42 AM, Larry Marshall wrote: > In the http://wiki.freeswitch.org/wiki/Getting_Started_Guide under Basic > Calling, it states ?Call your own extension number to login to your > voicemail box?. When I dial my extension (1000) from an internal softphone > at the same extension, it just asks me to leave message. It does not give me > the opportunity to hear my messages. > > > > What am I missing. I am using the default configuration. > > > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/31c314de/attachment.html From jonny.voip at gmail.com Wed May 20 12:28:57 2009 From: jonny.voip at gmail.com (Jon DiVita) Date: Wed, 20 May 2009 15:28:57 -0400 Subject: [Freeswitch-users] Conference Statistics Message-ID: <851b216b0905201228r78405eb0mefaae5ddd083200d@mail.gmail.com> Hello, all. Being new to FS, I was curious if there are any logs/cdrs which could be generated to gather statistics about a conference call? I'm mainly looking for call duration and user count. So far, my CDR's only have individual user CDR's, but nothing for a conference bridge. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/d9ebcf23/attachment.html From larclap at yahoo.com Wed May 20 12:54:52 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 20 May 2009 12:54:52 -0700 Subject: [Freeswitch-users] Retrieving voicemail In-Reply-To: <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> References: <001801c9d972$6272ce40$27586ac0$@com> <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> Message-ID: <004701c9d984$d7a03100$86e09300$@com> Thanks, Michael. I should have looked at conf/dialplan/default.xml. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 20, 2009 12:04 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Retrieving voicemail Lars, Thanks for pointing this out. I will update the wiki. The new way to check voicemail is to dial 4000 and then enter your extension. -MC On Wed, May 20, 2009 at 10:42 AM, Larry Marshall wrote: In the http://wiki.freeswitch.org/wiki/Getting_Started_Guide under Basic Calling, it states "Call your own extension number to login to your voicemail box". When I dial my extension (1000) from an internal softphone at the same extension, it just asks me to leave message. It does not give me the opportunity to hear my messages. What am I missing. I am using the default configuration. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/8a119825/attachment.html From lon at kickasspixels.com Wed May 20 14:47:54 2009 From: lon at kickasspixels.com (Lon Baker) Date: Wed, 20 May 2009 14:47:54 -0700 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? Message-ID: <5d3e0dc60905201447y6d894facja65c2504fb578a86@mail.gmail.com> When I issue a fsctl shutdown via xmlrpc I get a segmentation fault on Ubuntu server 9. It a 32 bit version with all packages fully up to date running freeswitch 1.0.3 Any ideas? Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/4c07091b/attachment.html From brian at freeswitch.org Wed May 20 15:01:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 20 May 2009 17:01:45 -0500 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? In-Reply-To: <5d3e0dc60905201447y6d894facja65c2504fb578a86@mail.gmail.com> References: <5d3e0dc60905201447y6d894facja65c2504fb578a86@mail.gmail.com> Message-ID: I would update to svn trunk and try again! ;) /b On May 20, 2009, at 4:47 PM, Lon Baker wrote: > When I issue a fsctl shutdown via xmlrpc I get a segmentation fault > on Ubuntu server 9. > > It a 32 bit version with all packages fully up to date running > freeswitch 1.0.3 > > Any ideas? > > Lon > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090520/aab355ad/attachment-0001.html From jason at jasonjgw.net Wed May 20 17:16:41 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 21 May 2009 10:16:41 +1000 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? In-Reply-To: <5d3e0dc60905201447y6d894facja65c2504fb578a86@mail.gmail.com> References: <5d3e0dc60905201447y6d894facja65c2504fb578a86@mail.gmail.com> Message-ID: <20090521001641.GA7284@jdc.jasonjgw.net> Lon Baker wrote: > When I issue a fsctl shutdown via xmlrpc I get a segmentation fault on > Ubuntu server 9. I think there was a fix to fsctl to eliminate segfaults recently. If you upgrade to trunk it might work now. From lcm at marshap.com Thu May 21 09:22:36 2009 From: lcm at marshap.com (Larry Marshall) Date: Thu, 21 May 2009 09:22:36 -0700 Subject: [Freeswitch-users] Documentation error? Message-ID: <006901c9da30$5a431180$0ec93480$@com> On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under Configuration, it speaks about the vars.xml file. Specifically it states, "In this file, there is only one parameter that you need to specify. That parameter is $${sip_profile}." I can't find the variable, nor can I grep for its assignment in conf. Am I missing something? Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/50dbca0b/attachment.html From brian at freeswitch.org Thu May 21 09:31:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 11:31:57 -0500 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <006901c9da30$5a431180$0ec93480$@com> References: <006901c9da30$5a431180$0ec93480$@com> Message-ID: <6BAF59D3-23E9-4C9D-BE20-D0CF083100CA@freeswitch.org> Its an error on the wiki you should have $${domain} in there /b On May 21, 2009, at 11:22 AM, Larry Marshall wrote: > On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under > Configuration, it speaks about the vars.xml file. Specifically it > states, ?In this file, there is only one parameter that you need to > specify. That parameter is $${sip_profile}.? > > I can?t find the variable, nor can I grep for its assignment in > conf. Am I missing something? > > Lars > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/7feef79e/attachment.html From gmaruzz at celliax.org Thu May 21 09:44:04 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 21 May 2009 18:44:04 +0200 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <6BAF59D3-23E9-4C9D-BE20-D0CF083100CA@freeswitch.org> References: <006901c9da30$5a431180$0ec93480$@com> <6BAF59D3-23E9-4C9D-BE20-D0CF083100CA@freeswitch.org> Message-ID: <7b197bef0905210944n184486b0w65e0d51a8effa5f6@mail.gmail.com> fixed On Thu, May 21, 2009 at 6:31 PM, Brian West wrote: > Its an error on the wiki you should have $${domain} in there > /b > On May 21, 2009, at 11:22 AM, Larry Marshall wrote: > > On the page?http://wiki.freeswitch.org/wiki/Configuring_SIP?under > Configuration, it speaks about the vars.xml file. Specifically it states, > ?In this file, there is only one parameter that you need to specify. That > parameter is $${sip_profile}.? > > I can?t find the variable, nor can I grep for its assignment in conf. Am I > missing something? > > Lars > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu May 21 11:06:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 May 2009 11:06:26 -0700 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <7b197bef0905210944n184486b0w65e0d51a8effa5f6@mail.gmail.com> References: <006901c9da30$5a431180$0ec93480$@com> <6BAF59D3-23E9-4C9D-BE20-D0CF083100CA@freeswitch.org> <7b197bef0905210944n184486b0w65e0d51a8effa5f6@mail.gmail.com> Message-ID: <87f2f3b90905211106j35f3c5c7t8c122e1a67bf84db@mail.gmail.com> Thanks for catching those little glitches!!! -MC On Thu, May 21, 2009 at 9:44 AM, Giovanni Maruzzelli wrote: > fixed > > > > On Thu, May 21, 2009 at 6:31 PM, Brian West wrote: > > Its an error on the wiki you should have $${domain} in there > > /b > > On May 21, 2009, at 11:22 AM, Larry Marshall wrote: > > > > On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under > > Configuration, it speaks about the vars.xml file. Specifically it states, > > ?In this file, there is only one parameter that you need to specify. That > > parameter is $${sip_profile}.? > > > > I can?t find the variable, nor can I grep for its assignment in conf. Am > I > > missing something? > > > > Lars > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > Brian West > > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/256e3cb0/attachment.html From freeswitch at cartissolutions.com Thu May 21 13:17:21 2009 From: freeswitch at cartissolutions.com (Yossi Neiman) Date: Thu, 21 May 2009 15:17:21 -0500 Subject: [Freeswitch-users] Conference Statistics In-Reply-To: <851b216b0905201228r78405eb0mefaae5ddd083200d@mail.gmail.com> References: <851b216b0905201228r78405eb0mefaae5ddd083200d@mail.gmail.com> Message-ID: <4A15B6D1.80003@cartissolutions.com> I don't *believe* there is anything of this nature built in at this time. I would imagine it can be implemented. If you yourself don't know C, but would like this functionality, three ways to get something like this added would be: A) Put up a bounty on the bounty page of the Wiki and hope somebody takes interest in doing it in exchange for a sum of money. B) Directly hire the core development team or a consultant to do the work. C) Put up a RFE (request for enhancement) on JIRA under the conference module. (Option A is more likely to motivate folks, and option B is the fastest way to get it done.) Now that I've said that I don't believe it currently has these features, I'm waiting for Brian West to come in and correct me like he usually does. :-) -- Yossi Neiman Cartis Solutions, Inc. http://www.cartissolutions.com Jon DiVita wrote: > Hello, all. Being new to FS, I was curious if there are any logs/cdrs > which could be generated to gather statistics about a conference > call? I'm mainly looking for call duration and user count. So far, > my CDR's only have individual user CDR's, but nothing for a conference > bridge. > > Thanks! From brian at freeswitch.org Thu May 21 13:26:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 15:26:35 -0500 Subject: [Freeswitch-users] Conference Statistics In-Reply-To: <4A15B6D1.80003@cartissolutions.com> References: <851b216b0905201228r78405eb0mefaae5ddd083200d@mail.gmail.com> <4A15B6D1.80003@cartissolutions.com> Message-ID: <0C18A731-DC43-41A9-8201-DBC85E6AFF29@freeswitch.org> No, this doesn't exist yet in FreeSWITCH but I do like the idea. /b On May 21, 2009, at 3:17 PM, Yossi Neiman wrote: > Now that I've said that I don't believe it currently has these > features, > I'm waiting for Brian West to come in and correct me like he usually > does. :-) Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/4ccb0422/attachment-0001.html From lon at kickasspixels.com Thu May 21 13:29:08 2009 From: lon at kickasspixels.com (Lon Baker) Date: Thu, 21 May 2009 13:29:08 -0700 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? Message-ID: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> I updates to the latest trunk this morning and still get the same segmentation issue on 32 bit Ubuntu server 9. The exact command I am issuing is fsctl shutdown asap. Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/b95308c2/attachment.html From brian at freeswitch.org Thu May 21 13:34:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 15:34:10 -0500 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? In-Reply-To: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> References: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> Message-ID: <520F5F16-A683-4993-9FD7-483FA3F50B58@freeswitch.org> prefix it with bgapi fsctl shutdown You can't do a shut down and wait for it to stop and give you a response because XML RPC is blocking. /b On May 21, 2009, at 3:29 PM, Lon Baker wrote: > I updates to the latest trunk this morning and still get the same > segmentation issue on 32 bit Ubuntu server 9. > > The exact command I am issuing is fsctl shutdown asap. > > Lon Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/f7d20c74/attachment.html From brian at freeswitch.org Thu May 21 13:34:19 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 15:34:19 -0500 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? In-Reply-To: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> References: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> Message-ID: Also open a jira please. /b On May 21, 2009, at 3:29 PM, Lon Baker wrote: > I updates to the latest trunk this morning and still get the same > segmentation issue on 32 bit Ubuntu server 9. > > The exact command I am issuing is fsctl shutdown asap. > > Lon Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/80cf1af3/attachment.html From brian at freeswitch.org Thu May 21 13:59:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 15:59:53 -0500 Subject: [Freeswitch-users] Segmentation fault with xmlrpc shutdown? In-Reply-To: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> References: <5d3e0dc60905211329n2b431c5cs83d509c7f64451d1@mail.gmail.com> Message-ID: <75B7073F-4F74-4EDC-BA3B-CDFAB87CC52F@freeswitch.org> We just tested this on our 64bit system and it works fine. Can you open a jira with a backtrace attached? Also how did you update? /b On May 21, 2009, at 3:29 PM, Lon Baker wrote: > I updates to the latest trunk this morning and still get the same > segmentation issue on 32 bit Ubuntu server 9. > > The exact command I am issuing is fsctl shutdown asap. > > Lon > ____________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/55724d02/attachment.html From larclap at yahoo.com Thu May 21 16:56:28 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 21 May 2009 16:56:28 -0700 Subject: [Freeswitch-users] Inbound call routing help Message-ID: <004a01c9da6f$c1e40120$45ac0360$@com> I want to setup a dialplan for a single DID. I would like it to go to a specific extension, and if not picked up in 15 seconds, go to voicemail. I have set this scenario up and it works. But I would also like this person to be able to call this DID from outside FS via a phone and be able to retrieve their voicemail. I've seen the example of how to pick up an extension's voicemail while inside FS by checking to see if the destination_number is the same as the caller_id_number, and if so, listen to voicemail, otherwise leave the message with voicemail. But I don't have a clue how to accomplish this from outside, other than dedicating another DID to solely retrieving voicemail from outside. Any ideas? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/453eacdf/attachment.html From brian at freeswitch.org Thu May 21 17:01:03 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 19:01:03 -0500 Subject: [Freeswitch-users] Inbound call routing help In-Reply-To: <004a01c9da6f$c1e40120$45ac0360$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> Message-ID: Try pressing * during the greeting and make sure you have the vmain extension so you can login. /b On May 21, 2009, at 6:56 PM, Lars Zeb wrote: > I want to setup a dialplan for a single DID. I would like it to go > to a specific extension, and if not picked up in 15 seconds, go to > voicemail. > > I have set this scenario up and it works. But I would also like this > person to be able to call this DID from outside FS via a phone and > be able to retrieve their voicemail. I?ve seen the example of how to > pick up an extension?s voicemail while inside FS by checking to see > if the destination_number is the same as the caller_id_number, and > if so, listen to voicemail, otherwise leave the message with > voicemail. > > But I don?t have a clue how to accomplish this from outside, other > than dedicating another DID to solely retrieving voicemail from > outside. > > Any ideas? > > Thanks, Lars > ______________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/8fb2e113/attachment-0001.html From daletrub at gmail.com Thu May 21 19:15:20 2009 From: daletrub at gmail.com (Dale Trub) Date: Thu, 21 May 2009 19:15:20 -0700 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines Message-ID: We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729 We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop. I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later). I'm attaching two different snippets of the FS log files where these issues are occurring. Does anyone have any recommendations about how to troubleshoot this? Any known issues/patches in FS that could be biting us? Is there some SIP logging we can do to debug? Are there any paid contractors avail who would have the expertise to look into this? Any help appreciated ... this is a major issue for us! Thanks much, -Dale daletrub at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/59704904/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.hangup Type: application/octet-stream Size: 11485 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/59704904/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.kick Type: application/octet-stream Size: 11951 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/59704904/attachment-0003.obj From diego.viola at gmail.com Thu May 21 19:23:56 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 21 May 2009 22:23:56 -0400 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: Message-ID: <86a32abc0905211923s12a47faw9482fe3c29c7b38e@mail.gmail.com> I have experienced the same a while ago, I originated calls from my freeswitch server to some landlines and calls would simply drop after X minutes. I tried to debug the thing but found nothing relevant, maybe I had the same issue as you. Let me know if you figure it out what it was. Diego On Thu, May 21, 2009 at 10:15 PM, Dale Trub wrote: > We're running FreeSwitch as part of a teleconferencing service, inside a > telcom?(so no > internet latency/NAT issues)?and using g.729 > We are?receiving some complaints of dropped calls, > including from landlines. ? This means they join the conference, and x > minutes in they simply drop. > I?know that cellphones tend to drop calls frequently, but landlines > are pretty reliable, and we're hearing it a lot. ?From the FreeSwitch side > of things, it just > looks like those callers hung up (but then dialed back in just a moment > later). > I'm attaching two different snippets of the FS log files where these issues > are occurring. > > Does anyone have any recommendations about how to troubleshoot this? > Any known issues/patches in FS that could be biting us? > Is there some SIP logging we can do to debug? > Are there any paid contractors avail who would have the expertise to look > into this? > Any help appreciated ... this is a major issue for us! > Thanks much, > -Dale > daletrub at gmail.com > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu May 21 19:28:21 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 21:28:21 -0500 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: Message-ID: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> On May 21, 2009, at 9:15 PM, Dale Trub wrote: > We're running FreeSwitch as part of a teleconferencing service, > inside a telcom (so no > internet latency/NAT issues) and using g.729 So you're using g729 with conferences? > We are receiving some complaints of dropped calls, > including from landlines. This means they join the conference, and > x minutes in they simply drop. > > I know that cellphones tend to drop calls frequently, but landlines > are pretty reliable, and we're hearing it a lot. From the > FreeSwitch side of things, it just > looks like those callers hung up (but then dialed back in just a > moment later). > > I'm attaching two different snippets of the FS log files where these > issues are occurring. Next time please call them .txt because you cause extra work to have to open them otherwise. > Does anyone have any recommendations about how to troubleshoot this? > > Any known issues/patches in FS that could be biting us? Depends you failed to include some very valid info such as what version or svn rev you're running and what linux distro. > Is there some SIP logging we can do to debug? Yes covered on the wiki. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch > Are there any paid contractors avail who would have the expertise to > look into this? email consulting at freeswitch.org > Any help appreciated ... this is a major issue for us! > > Thanks much, > > -Dale > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/73886e91/attachment.html From jason at jasonjgw.net Thu May 21 19:36:55 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 22 May 2009 12:36:55 +1000 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: Message-ID: <20090522023655.GA31114@jdc.jasonjgw.net> Dale Trub wrote: > Does anyone have any recommendations about how to troubleshoot this? sofia profile external siptrace on and watch the SIP traces to see what happens. > > Any known issues/patches in FS that could be biting us? You didn't say which version you were running. Does the problem still occur with the latest version from svn trunk? I assume there is a SIP service provider somewhere between your FreeSWITCH system and the PSTN. It is quite possible that the problem is on their end, especially if their system is terminating the calls. From the logs, it just looks like a standard hangup, which leads me to wonder whether the other side is responsible. Collecting SIP traces would seem to be a good idea. Disclaimer: I'm not an expert, and I'm sure there are paid contractors in the community who will gladly help, for a fee. From daletrub at gmail.com Thu May 21 20:09:22 2009 From: daletrub at gmail.com (Dale Trub) Date: Thu, 21 May 2009 20:09:22 -0700 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> Message-ID: Thanks Brian! To answer your questions: Freeswitch svn revision: 12148 Centos rev: 2.6.18-92.el5 And apologies, actually I guess we're using g711 not 729. Jason: I agree it would seem to be on the switch/telco side. And, the telco says many other people are in the same set-up as us and don't have any issues, so they're insisting it's on our end. On Thu, May 21, 2009 at 7:28 PM, Brian West wrote: > > On May 21, 2009, at 9:15 PM, Dale Trub wrote: > > We're running FreeSwitch as part of a teleconferencing service, inside a > telcom (so no > internet latency/NAT issues) and using g.729 > > > So you're using g729 with conferences? > > We are receiving some complaints of dropped calls, > including from landlines. This means they join the conference, and x > minutes in they simply drop. > > I know that cellphones tend to drop calls frequently, but landlines > are pretty reliable, and we're hearing it a lot. From the FreeSwitch side > of things, it just > looks like those callers hung up (but then dialed back in just a moment > later). > > I'm attaching two different snippets of the FS log files where these issues > are occurring. > > > Next time please call them .txt because you cause extra work to have to > open them otherwise. > > Does anyone have any recommendations about how to troubleshoot this? > > Any known issues/patches in FS that could be biting us? > > > Depends you failed to include some very valid info such as what version or > svn rev you're running and what linux distro. > > Is there some SIP logging we can do to debug? > > > Yes covered on the wiki. > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch > > Are there any paid contractors avail who would have the expertise to look > into this? > > > email consulting at freeswitch.org > > Any help appreciated ... this is a major issue for us! > > Thanks much, > > -Dale > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/ec9a63b3/attachment-0001.html -------------- next part -------------- 2009-05-07 17:32:34 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/phonenumber at ipaddr entering state [terminated] 2009-05-07 17:32:34 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/external/phonenumber at ipaddr [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-07 17:32:34 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/phonenumber at ipaddr [KILL] 2009-05-07 17:32:34 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/external/phonenumber at ipaddr [BREAK] 2009-05-07 17:32:34 [DEBUG] mod_conference.c:2132 conference_loop_output() Channel leaving conference, cause: NORMAL_CLEARING 2009-05-07 17:32:34 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read_codec() Restore original codec. 2009-05-07 17:32:34 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/external/phonenumber at ipaddr) State EXECUTE going to sleep 2009-05-07 17:32:34 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/external/phonenumber at ipaddr) Running State Change CS_HANGUP 2009-05-07 17:32:34 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/phonenumber at ipaddr) State HANGUP 2009-05-07 17:32:34 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/external/phonenumber at ipaddr Overriding SIP cause 480 with 200 from the other leg 2009-05-07 17:32:34 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/phonenumber at ipaddr hanging up, cause: NORMAL_CLEARING 2009-05-07 17:32:34 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/phonenumber at ipaddr Standard HANGUP, cause: NORMAL_CLEARING 2009-05-07 17:32:34 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/phonenumber at ipaddr) State HANGUP going to sleep 2009-05-07 17:32:34 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 1824 (sofia/external/phonenumber at ipaddr) Locked, Waiting on external entities 2009-05-07 17:32:34 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1824 (sofia/external/phonenumber at ipaddr) Ended 2009-05-07 17:32:34 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/phonenumber at ipaddr [CS_HANGUP] 2009-05-07 17:32:34 [DEBUG] mod_conference.c:1255 conference_thread_run() Write Lock ON 2009-05-07 17:32:34 [DEBUG] mod_conference.c:2290 conference_record_thread_run() Recording Stopped 2009-05-07 17:32:34 [DEBUG] mod_conference.c:1258 conference_thread_run() Write Lock OFF 2009-05-07 17:32:34 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal) 2009-05-07 17:32:34 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (0 callers)','','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Active (0 callers)] event_type: [presence] alt_event_type: [dialog] event_count: [1] unique-id: [3000] channel-state: [CS_ROUTING] answer-state: [confirmed] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:32:34] Event-Date-GMT: [Fri, 08 May 2009 00:32:34 GMT] Event-Date-Timestamp: [1241742754401218] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_del_member] Event-Calling-Line-Number: [769] 2009-05-07 17:32:34 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2009-05-07 17:32:34 [WARNING] sofia_presence.c:531 actual_sofia_presence_event_handler() external is passive, skipping 2009-05-07 17:32:34 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:32:34 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (0 callers)','','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Active (0 callers)] event_type: [presence] alt_event_type: [dialog] event_count: [1] unique-id: [3000] channel-state: [CS_ROUTING] answer-state: [confirmed] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:32:34] Event-Date-GMT: [Fri, 08 May 2009 00:32:34 GMT] Event-Date-Timestamp: [1241742754401218] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_del_member] Event-Calling-Line-Number: [769] 2009-05-07 17:32:34 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() default is an alias, skipping 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() nat is an alias, skipping 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() ipaddr is an alias, skipping 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() outbound is an alias, skipping 2009-05-07 17:32:34 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal) 2009-05-07 17:32:34 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Inactive','busy','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Inactive] rpid: [idle] event_type: [presence] alt_event_type: [dialog] event_count: [2] unique-id: [3000] channel-state: [CS_HANGUP] answer-state: [terminated] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:32:34] Event-Date-GMT: [Fri, 08 May 2009 00:32:34 GMT] Event-Date-Timestamp: [1241742754421218] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_thread_run] Event-Calling-Line-Number: [1184] 2009-05-07 17:32:34 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2009-05-07 17:32:34 [WARNING] sofia_presence.c:531 actual_sofia_presence_event_handler() external is passive, skipping 2009-05-07 17:32:34 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:32:34 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Inactive','busy','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Inactive] rpid: [idle] event_type: [presence] alt_event_type: [dialog] event_count: [2] unique-id: [3000] channel-state: [CS_HANGUP] answer-state: [terminated] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:32:34] Event-Date-GMT: [Fri, 08 May 2009 00:32:34 GMT] Event-Date-Timestamp: [1241742754421218] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_thread_run] Event-Calling-Line-Number: [1184] 2009-05-07 17:32:34 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() default is an alias, skipping 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() nat is an alias, skipping 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() ipaddr is an alias, skipping 2009-05-07 17:32:34 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() outbound is an alias, skipping -------------- next part -------------- 2009-05-07 17:36:28 [DEBUG] mod_conference.c:2950 conf_api_sub_kick() Send signal sofia/external/phonenumber at ipaddr [BREAK] 2009-05-07 17:36:28 [DEBUG] mod_conference.c:2132 conference_loop_output() Channel leaving conference, cause: NONE 2009-05-07 17:36:28 [DEBUG] mod_conference.c:5006 conference_function() sofia/external/phonenumber at ipaddr receive message [UNBRIDGE] 2009-05-07 17:36:28 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/external/phonenumber at ipaddr [BREAK] 2009-05-07 17:36:28 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read_codec() Restore original codec. 2009-05-07 17:36:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-05-07 17:36:28 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal) 2009-05-07 17:36:28 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (0 callers)','','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Active (0 callers)] event_type: [presence] alt_event_type: [dialog] event_count: [4] unique-id: [3000] channel-state: [CS_ROUTING] answer-state: [confirmed] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:36:28] Event-Date-GMT: [Fri, 08 May 2009 00:36:28 GMT] Event-Date-Timestamp: [1241742988481266] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_del_member] Event-Calling-Line-Number: [769] 2009-05-07 17:36:28 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2009-05-07 17:36:28 [WARNING] sofia_presence.c:531 actual_sofia_presence_event_handler() external is passive, skipping 2009-05-07 17:36:28 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:36:28 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (0 callers)','','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Active (0 callers)] event_type: [presence] alt_event_type: [dialog] event_count: [4] unique-id: [3000] channel-state: [CS_ROUTING] answer-state: [confirmed] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:36:28] Event-Date-GMT: [Fri, 08 May 2009 00:36:28 GMT] Event-Date-Timestamp: [1241742988481266] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_del_member] Event-Calling-Line-Number: [769] 2009-05-07 17:36:28 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() default is an alias, skipping 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() nat is an alias, skipping 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() ipaddr is an alias, skipping 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() outbound is an alias, skipping 2009-05-07 17:36:28 [DEBUG] mod_conference.c:1255 conference_thread_run() Write Lock ON 2009-05-07 17:36:28 [DEBUG] mod_conference.c:2290 conference_record_thread_run() Recording Stopped 2009-05-07 17:36:28 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal) 2009-05-07 17:36:28 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Inactive','busy','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Inactive] rpid: [idle] event_type: [presence] alt_event_type: [dialog] event_count: [5] unique-id: [3000] channel-state: [CS_HANGUP] answer-state: [terminated] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:36:28] Event-Date-GMT: [Fri, 08 May 2009 00:36:28 GMT] Event-Date-Timestamp: [1241742988501266] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_thread_run] Event-Calling-Line-Number: [1184] 2009-05-07 17:36:28 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2009-05-07 17:36:28 [WARNING] sofia_presence.c:531 actual_sofia_presence_event_handler() external is passive, skipping 2009-05-07 17:36:28 [INFO] sofia_presence.c:561 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:36:28 [ERR] sofia_presence.c:569 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Inactive','busy','ipaddr',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='3000' and (sub_to_host='ipaddr' or presence_hosts like '%ipaddr%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: proto: [conf] login: [3000] from: [3000 at ipaddr] status: [Inactive] rpid: [idle] event_type: [presence] alt_event_type: [dialog] event_count: [5] unique-id: [3000] channel-state: [CS_HANGUP] answer-state: [terminated] call-direction: [inbound] Event-Name: [PRESENCE_IN] Core-UUID: [31042480-7d5b-4d01-bda0-ef5758d51085] FreeSWITCH-Hostname: [servername] FreeSWITCH-IPv4: [ipaddr] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-07 17:36:28] Event-Date-GMT: [Fri, 08 May 2009 00:36:28 GMT] Event-Date-Timestamp: [1241742988501266] Event-Calling-File: [mod_conference.c] Event-Calling-Function: [conference_thread_run] Event-Calling-Line-Number: [1184] 2009-05-07 17:36:28 [INFO] sofia_presence.c:577 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal-ipv6) 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() default is an alias, skipping 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() nat is an alias, skipping 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() ipaddr is an alias, skipping 2009-05-07 17:36:28 [WARNING] sofia_presence.c:524 actual_sofia_presence_event_handler() outbound is an alias, skipping 2009-05-07 17:36:30 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-05-07 17:36:30 [DEBUG] mod_conference.c:1258 conference_thread_run() Write Lock OFF 2009-05-07 17:36:30 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/phonenumber at ipaddr [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-07 17:36:30 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/phonenumber at ipaddr [KILL] 2009-05-07 17:36:30 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/external/phonenumber at ipaddr [BREAK] 2009-05-07 17:36:30 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/external/phonenumber at ipaddr) State EXECUTE going to sleep 2009-05-07 17:36:30 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/external/phonenumber at ipaddr) Running State Change CS_HANGUP 2009-05-07 17:36:30 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/phonenumber at ipaddr) State HANGUP 2009-05-07 17:36:30 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/phonenumber at ipaddr hanging up, cause: NORMAL_CLEARING 2009-05-07 17:36:30 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/external/phonenumber at ipaddr 2009-05-07 17:36:30 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/phonenumber at ipaddr Standard HANGUP, cause: NORMAL_CLEARING 2009-05-07 17:36:30 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/external/phonenumber at ipaddr) State HANGUP going to sleep 2009-05-07 17:36:30 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 1825 (sofia/external/phonenumber at ipaddr) Locked, Waiting on external entities 2009-05-07 17:36:30 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1825 (sofia/external/phonenumber at ipaddr) Ended 2009-05-07 17:36:30 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/phonenumber at ipaddr [CS_HANGUP] From jason at jasonjgw.net Thu May 21 20:39:19 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 22 May 2009 13:39:19 +1000 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> Message-ID: <20090522033919.GA5309@jdc.jasonjgw.net> Dale Trub wrote: > Freeswitch svn revision: 12148 It's time for an upgrade. The developers have fixed a lot of bugs since then. From msc at freeswitch.org Thu May 21 20:54:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 May 2009 20:54:24 -0700 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required Message-ID: FYI, We just want to let everyone know that we have made a few updates that will require a rebootstrap. One of the key updates was a security fix for libsndfile. In this particular case it won't be possible simply to "make current" like you normally do. Here is a common set of commands for a typical Linux rebootstrap: cd /usr/src/freeswitch.trunk make clean svn up ./bootstrap.sh ./configure make install NOTE: if you've got the libzrtp file and you've already run the buildzrtp.sh script then be sure to use "./configure --enable-zrtp" in the above operation. Thank you for your continued support of the FreeSWITCH project! -Michael S Collins http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/a621feed/attachment.html From jim at evolutiontel.net Thu May 21 21:17:07 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 22 May 2009 14:17:07 +1000 Subject: [Freeswitch-users] Secure RTP Message-ID: Hi All, Have been trying to workout how to solve a call scenario involving SRTP and need some help. The scenario is: Eyebeam ---->FS----->Eyebeam with make and accept only encrypted calls set. What I am hoping to acheive is, if the A leg does not have SRTP set and no SRTP Descriptors are sent in the INVITE to the B leg, when the B leg responds with 415 Bad Security Level this is intercepted and a re-invite is sent with the security descriptions so this call, 1, terminates and 2, is B leg secure. In my dialplan I have CONTINUE_ON_FAILURE set for 79 and then set BYPASS_MEDIA=FALSE (was SET true earlier in the Dialplan) and EXPORT SIP_SECURE_MEDIA=TRUE then finaly bridge the call once more. What I get is Eyebeam FS Eyebeam ---->INVITE no SRTP----> <---100 Trying <--------- ---->INVITE no SRTP----> <---415 Bad Security<---- ---->INVITE no SRTP----> Based on the following debug snippet showing the continue on fail occuring, the Local Key is generated, however it is not added to the second INVITE message when sent out. 2009-05-22 13:23:40 [DEBUG] switch_ivr_originate.c:2094 switch_ivr_originate() Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2009-05-22 13:23:40 [INFO] mod_dptools.c:2074 audio_bridge_function() Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED 2009-05-22 13:23:40 [DEBUG] mod_dptools.c:2101 audio_bridge_function() Continue on fail [true]: Cause: SERVICE_NOT_IMPLEMENTED EXECUTE sofia/internal/0631000002 at sip.evolutiontel.net set(bypass_media=false) 2009-05-22 13:23:40 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/0631000002 at sip.evolutiontel.net SET [bypass_media]=[false] EXECUTE sofia/internal/0631000002 at sip.evolutiontel.net export(sip_secure_media=true) 2009-05-22 13:23:40 [DEBUG] mod_dptools.c:886 export_function() EXPORT [sip_secure_media]=[true] EXECUTE sofia/internal/0631000002 at sip.evolutiontel.net export(sip_crypto_mandatory=true) 2009-05-22 13:23:40 [DEBUG] mod_dptools.c:886 export_function() EXPORT [sip_crypto_mandatory]=[true] EXECUTE sofia/internal/0631000002 at sip.evolutiontel.net bridge({sip_from_uri=sip:0631000002 at sip.evolutiontel.net}sofia/sip.evolutiontel.net/0631000006 at 192.168.0.3^103 at sip.evolutiontel.net) 2009-05-22 13:23:40 [DEBUG] switch_ivr_originate.c:1082 switch_ivr_originate() variable string 0 = [sip_from_uri=sip:0631000002 at sip.evolutiontel.net] 2009-05-22 13:23:40 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/0631000006 at 192.168.0.3 [b0e7a29c-b154-4b9c-b851-4d987669571f] 2009-05-22 13:23:40 [DEBUG] mod_sofia.c:2659 sofia_outgoing_channel() (sofia/internal/0631000006 at 192.168.0.3) State Change CS_NEW -> CS_INIT 2009-05-22 13:23:40 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000006 at 192.168.0.3 [BREAK] 2009-05-22 13:23:40 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 225 (sofia/internal/0631000006 at 192.168.0.3) Ended 2009-05-22 13:23:40 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/0631000006 at 192.168.0.3 [CS_DESTROY] 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0631000006 at 192.168.0.3) State DESTROY 2009-05-22 13:23:40 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/internal/0631000006 at 192.168.0.3 SOFIA DESTROY 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/0631000006 at 192.168.0.3 Standard DESTROY 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/0631000006 at 192.168.0.3) State DESTROY going to sleep 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) Running State Change CS_INIT 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) State INIT 2009-05-22 13:23:40 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/0631000006 at 192.168.0.3 SOFIA INIT 2009-05-22 13:23:40 [DEBUG] sofia_glue.c:1972 sofia_glue_build_crypto() Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:0mENEM7ab0d6DtmcCMgDmbHXlYIHpdXdLnVCtFYX] 2009-05-22 13:23:40 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/0631000006 at 192.168.0.3) State Change CS_INIT -> CS_ROUTING 2009-05-22 13:23:40 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000006 at 192.168.0.3 [BREAK] 2009-05-22 13:23:40 [DEBUG] sofia.c:2911 sofia_handle_sip_i_state() Channel sofia/internal/0631000006 at 192.168.0.3 entering state [calling][0] 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) State INIT going to sleep 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) Running State Change CS_ROUTING 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) State ROUTING 2009-05-22 13:23:40 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/0631000006 at 192.168.0.3 SOFIA ROUTING 2009-05-22 13:23:40 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/0631000006 at 192.168.0.3) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-05-22 13:23:40 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/0631000006 at 192.168.0.3 [BREAK] 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) State ROUTING going to sleep 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) Running State Change CS_CONSUME_MEDIA 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) State CONSUME_MEDIA 2009-05-22 13:23:40 [DEBUG] switch_core_state_machine.c:502 switch_core_session_run() (sofia/internal/0631000006 at 192.168.0.3) State CONSUME_MEDIA going to sleep 2009-05-22 13:23:40 [DEBUG] sofia.c:2911 sofia_handle_sip_i_state() Channel sofia/internal/0631000006 at 192.168.0.3 entering state [terminated][415] If I swap and A leg is set to have SRTP and B party does not it will be setup with only the A leg having SRTP and the B leg being normal RTP. Let me know if you need any further info, or if this is a fault and will need a jira ticket opened. FreeSWITCH Version 1.0.trunk (13232) Thanks, Jim From diego.viola at gmail.com Thu May 21 21:26:51 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 22 May 2009 00:26:51 -0400 Subject: [Freeswitch-users] Cool names for my VoIP company Message-ID: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> Hey guys, I'm about to start my own ITSP with FreeSWITCH, and I'm looking some cool names for my VoIP company, if you know some please tell me :) Diego From brian at freeswitch.org Thu May 21 21:44:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 23:44:08 -0500 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: Message-ID: Jim, You seem to be making the whole ordeal overly complex for no reason. You can not accept the call and send it out and get a 415 back and expect to do the process all over again automatically. Once you get the 415 its done.. finished OVER gotta do it again. Remember we are a B2BUA not a proxy... so its all done per leg. If all else fails just go get libzrtp and install it as per the wiki and enable zrtp support. Then get zfone for your two PC's and you're done. We don't do the trusted man in the middle stuff yet but you can get the same end result. /b On May 21, 2009, at 11:17 PM, Jim Burke wrote: > What I am hoping to acheive is, if the A leg does not have SRTP set > and no SRTP Descriptors are sent in the INVITE to the B leg, when the > B leg responds with 415 Bad Security Level this is intercepted and a > re-invite is sent with the security descriptions so this call, 1, > terminates and 2, is B leg secure. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/57e5abd5/attachment.html From brian at freeswitch.org Thu May 21 21:47:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 21 May 2009 23:47:48 -0500 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> Message-ID: <48245F1F-EE66-4CEC-A0A6-AF5F1DFA3CD9@freeswitch.org> If I had a few bucks for every time the telco has said this to me I could just about retire! You using 100% SIP? /b On May 21, 2009, at 10:09 PM, Dale Trub wrote: > And, the telco says many other people are in the same set-up as us > and don't have any issues, so they're insisting it's on our end. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/9746f152/attachment-0001.html From sridhart at alcatel-lucent.com Thu May 21 22:24:34 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Fri, 22 May 2009 10:54:34 +0530 Subject: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy Message-ID: <9389DD3DDD6B9144B147CE564C6599B9041D86B8BD@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, I want to use FreeSWITCH as a SIP transparent proxy in session border controller application. Please let me know the changes in configuration files required to achieve this behaviour Thanks very much for the help. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/0006df56/attachment.html From mattdfong at gmail.com Thu May 21 22:44:18 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 21 May 2009 22:44:18 -0700 Subject: [Freeswitch-users] Best way to determine if a bridge was successful in Lua Message-ID: <4256bf830905212244o13fd1582v25ba125185f931fd@mail.gmail.com> I'm using a lua script to control an IVR, and would like to know how I can tell if a session:execute("bridge","sofia/gateway/blahblah"); was successful or not it seems the response from session:execute is nil regardless if the bridge was successful or not whats the best way? Thanks --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/6d28e4f9/attachment.html From jim at evolutiontel.net Thu May 21 22:47:43 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 22 May 2009 15:47:43 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: Message-ID: Hey Brian, Will have a look at ZRTP :) Not sure I understand your comments regarding its all over once receiving the 415 from the B party. Is'nt that what parm continue_on_fail does? The fact that it sends the invite back out sorta proves this. The other point of interest here is that if you set before the first bridge function it will include the security descriptions in the B leg INVITE even when the A leg does not have them and the call will succeed. The B Eyebeam will show the locked padlock while A does not. >From what I can see in code it is this guy that must stop it all from happening. TFLAG_SECURE But I dont understand why :( Regards, Jim On Fri, May 22, 2009 at 2:44 PM, Brian West wrote: > Jim, > You seem to be making the whole ordeal overly complex for no reason. > > ?? ? ? expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" > break="never"> > > ?? ? ? ? data="sip_secure_media=${sip_has_crypto}"/> > > ?? ? ? > You can not accept the call and send it out and get a 415 back and expect to > do the process all over again automatically. ?Once you get the 415 its > done.. finished OVER gotta do it again. ?Remember we are a B2BUA not a > proxy... so its all done per leg. > If all else fails just go get libzrtp and install it as per the wiki and > enable zrtp support. ?Then get zfone for your two PC's and you're done. ?We > don't do the trusted man in the middle stuff yet but you can get the same > end result. > /b > > > > On May 21, 2009, at 11:17 PM, Jim Burke wrote: > > What I am hoping to acheive is, if the A leg does not have SRTP set > and no SRTP Descriptors are sent in the INVITE to the B leg, when the > B leg responds with 415 Bad Security Level this is intercepted and a > re-invite is sent with the security descriptions so this call, 1, > terminates and 2, is B leg secure. > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mattdfong at gmail.com Thu May 21 23:09:21 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 21 May 2009 23:09:21 -0700 Subject: [Freeswitch-users] Best way to determine if a bridge was successful in Lua In-Reply-To: <4256bf830905212244o13fd1582v25ba125185f931fd@mail.gmail.com> References: <4256bf830905212244o13fd1582v25ba125185f931fd@mail.gmail.com> Message-ID: <4256bf830905212309k41b91660o9d443d854a837259@mail.gmail.com> hrm...it's also seems to be that if my lua script looks like session:execute("bridge", "sofia/gateway/XXX/0XXXXX") session:execute("bridge", "sofia/gateway/XXXX/XXX") if the first bridge fails, the session is immediately hungup, even if hangup_after_bridge is set to false...is this the intended behavior? I'm not trying to setup failover--I know I can use | to setup a bridge failover, but would like to retain use of the lua ivr script should a bridge fail. If I want to redirect to a voicemail or recorded message, on bridge fail, how can I do this? Thanks again. --matt On Thu, May 21, 2009 at 10:44 PM, Matthew Fong wrote: > I'm using a lua script to control an IVR, and would like to know how I can > tell if a > session:execute("bridge","sofia/gateway/blahblah"); > > was successful or not > > it seems the response from session:execute is nil regardless if the bridge > was successful or not > > whats the best way? Thanks > > --matt > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/90fbdc19/attachment.html From mattdfong at gmail.com Thu May 21 23:14:37 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 21 May 2009 23:14:37 -0700 Subject: [Freeswitch-users] Best way to determine if a bridge was successful in Lua In-Reply-To: <4256bf830905212309k41b91660o9d443d854a837259@mail.gmail.com> References: <4256bf830905212244o13fd1582v25ba125185f931fd@mail.gmail.com> <4256bf830905212309k41b91660o9d443d854a837259@mail.gmail.com> Message-ID: <4256bf830905212314l49ad5084sb12b4e225b413d50@mail.gmail.com> grr...continue_on_fail...ignore my ignorance ;) but it would still be nice getting a response back from the session:execute bridge --matt On Thu, May 21, 2009 at 11:09 PM, Matthew Fong wrote: > hrm...it's also seems to be that if my lua script looks like > session:execute("bridge", "sofia/gateway/XXX/0XXXXX") > session:execute("bridge", "sofia/gateway/XXXX/XXX") > > if the first bridge fails, the session is immediately hungup, even if > hangup_after_bridge is set to false...is this the intended behavior? > > I'm not trying to setup failover--I know I can use | to setup a bridge > failover, but would like to retain use of the lua ivr script should a bridge > fail. If I want to redirect to a voicemail or recorded message, on bridge > fail, how can I do this? Thanks again. > > --matt > > On Thu, May 21, 2009 at 10:44 PM, Matthew Fong wrote: > >> I'm using a lua script to control an IVR, and would like to know how I can >> tell if a >> session:execute("bridge","sofia/gateway/blahblah"); >> >> was successful or not >> >> it seems the response from session:execute is nil regardless if the bridge >> was successful or not >> >> whats the best way? Thanks >> >> --matt >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090521/aab67b9e/attachment.html From dujinfang at gmail.com Thu May 21 23:31:48 2009 From: dujinfang at gmail.com (seven) Date: Fri, 22 May 2009 14:31:48 +0800 Subject: [Freeswitch-users] Best way to determine if a bridge was successful in Lua In-Reply-To: <4256bf830905212314l49ad5084sb12b4e225b413d50@mail.gmail.com> References: <4256bf830905212244o13fd1582v25ba125185f931fd@mail.gmail.com> <4256bf830905212309k41b91660o9d443d854a837259@mail.gmail.com> <4256bf830905212314l49ad5084sb12b4e225b413d50@mail.gmail.com> Message-ID: <06FAAE54-DA70-4BC9-A3CA-9292E5F84A11@gmail.com> would you like to try this? bridge_hangup_cause = session:getVariable("bridge_hangup_cause") or session:getVariable("originate_disposition"); if (bridge_hangup_cause == "NORMAL_TEMPORARY_FAILURE" or bridge_hangup_cause == "NO_ROUTE_DESTINATION" or bridge_hangup_cause == "CALL_REJECTED") then -- blah... On May 22, 2009, at 2:14 PM, Matthew Fong wrote: > grr...continue_on_fail...ignore my ignorance ;) > > but it would still be nice getting a response back from the > session:execute bridge > > --matt > > On Thu, May 21, 2009 at 11:09 PM, Matthew Fong > wrote: > hrm...it's also seems to be that if my lua script looks like > > session:execute("bridge", "sofia/gateway/XXX/0XXXXX") > session:execute("bridge", "sofia/gateway/XXXX/XXX") > > if the first bridge fails, the session is immediately hungup, even > if hangup_after_bridge is set to false...is this the intended > behavior? > > I'm not trying to setup failover--I know I can use | to setup a > bridge failover, but would like to retain use of the lua ivr script > should a bridge fail. If I want to redirect to a voicemail or > recorded message, on bridge fail, how can I do this? Thanks again. > > --matt > > On Thu, May 21, 2009 at 10:44 PM, Matthew Fong > wrote: > I'm using a lua script to control an IVR, and would like to know how > I can tell if a > > session:execute("bridge","sofia/gateway/blahblah"); > > was successful or not > > it seems the response from session:execute is nil regardless if the > bridge was successful or not > > whats the best way? Thanks > > --matt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/39d3f26b/attachment.html From gavin.henry at gmail.com Fri May 22 00:31:04 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Fri, 22 May 2009 08:31:04 +0100 Subject: [Freeswitch-users] Retrieving voicemail In-Reply-To: <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> References: <001801c9d972$6272ce40$27586ac0$@com> <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> Message-ID: <13ca621c0905220031y76944fa2re3656d2c030e0577@mail.gmail.com> 2009/5/20 Michael Collins : > Lars, > > Thanks for pointing this out. I will update the wiki. The new way to check > voicemail is to dial 4000 and then enter your extension. What version is that on? I just followed http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and registered with 1000/1234 and then dialed 1000 which put me straight into voicemail. That link installs trunk, so should be the latest version and default config? Thanks. From saeedahmad1981 at gmail.com Fri May 22 02:01:17 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Fri, 22 May 2009 11:01:17 +0200 Subject: [Freeswitch-users] text to speech IVRs and MOH In-Reply-To: <4A12ECEB.1050207@gmx.net> References: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net> <4A12ECEB.1050207@gmx.net> Message-ID: Thanks guys for a detailed reply specially pete. On Tue, May 19, 2009 at 7:31 PM, Peter P GMX wrote: > Thanks for this overwiev. > > One question: How does this compare to Cepstral TTS? > > Best regards > Peter > > pete at privateconnect.com schrieb: > > I've spent the last 2-3 months on researching TTS and ASR for FS for a > > project. Best TTS depends on what you consider important. Also, how > > do you plan on using it. > > > > Here's some of the TTS engines I've run across with some pros/cons: > > > > Festivate Lite (flite) > > Pros: > > - Free (comes with FS) > > - simple to use > > - 16K voice sounds decent > > - Completely customizable > > Cons: > > - 8K voice sounds horrible over cell phone > > > > NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl > voice) > > Pros: > > - My selection for best soundig voices > > - Recently select by Stephen Hawkings for his voice (geek points!) > > - Lots of Languages supported > > - Free trial available > > Cons: > > - Custom C-Based API (FS interface coming soon) > > - Large file size (Engine + SDK + 1 Voice = 900MB) > > - Support is lacking (Company beed in Korean, time zone issues, etc) > > > > Nuance ($500/port for 1 voice) > > Pros: > > - Wide Variety of Products > > - Support MRCP > > - Supports ASR as well (add'l fees) > > - Excellent support > > - Free trial > > - Decent sounding voices at 8K and 16K > > - Wide range of tuning parameters > > Cons: > > - Pricey > > - Limited voice selection > > - Limited support for 64-bit linux > > > > AT&T (NaturalVoice) (no pricing info available) > > Pros: > > - Big company (solid in marketplace) > > - Good suppport (user and developer) > > - ASP model means no software to maintain > > Cons: > > - ASP model incurs delay > > - Voices sound too digitized > > - Limited support for 64-bit linux > > > > Loquendo ($500/port for 1 voice + 15% addl voice) > > Pros: > > - Good sounding voices (almost as good as NeoSpeech) > > - Wide variety of languages > > - Excellent support > > - Has free 30 day trial > > - Supports MRCP > > - Support ASR and Voice Recognition as well. (add'l fees) > > - Small footprint (< 150MB) > > Cons: > > - Pricey > > - Complicated install process > > - Limited management/tuning capabilities > > > > In the end, it was down to NeoSpeech or Loquendo for our application. > > We are currently running tests with NeoSpeech and assuming all goes > > well, we will select them. Though don't let that color your opinion > > too much after several "focus groups" we discovered the most important > > element in the equation is does your customer/boss like the sound of > > the voices, and that is a completely subjective decision. > > > > > > -pete > > > > -------- Original Message -------- > > Subject: [Freeswitch-users] text to speech IVRs and MOH > > From: Saeed Ahmad > > Date: Tue, May 19, 2009 12:40 am > > To: freeswitch-users at lists.freeswitch.org > > > > Hi all, > > > > Could you guys recommend me any online text to speech IVR software > > which works OK with FS. i am using AT&T site and for some IVRs i > > get sample rate errors. Also some resource to download more MOH > > wav files. > > > > Many thanks > > > ------------------------------------------------------------------------ > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/6e0b033d/attachment.html From oseslija at gmail.com Fri May 22 03:22:07 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 22 May 2009 12:22:07 +0200 Subject: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9041D86B8BD@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9041D86B8BD@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <4468a6770905220322q13609513k65ad14c7acfced6c@mail.gmail.com> Hello, FS by design is B2BUA, and it cannot route INVITEs and other SIP methods. It can however, bridge a-leg to b-leg with or w/o media and doing plenty other cool stuff much better than commercial projects. I suggest joining us on irc to detail your setup so we can help you. Regards, Ognjen (sekil on #freeswitch). On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi all, > > I want to use FreeSWITCH as a SIP transparent proxy in session border > controller application. Please let me know the changes in configuration > files required to achieve this behaviour > > Thanks very much for the help. > > Regards, > Sridhar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/2ed44fa4/attachment.html From Prometheus001 at gmx.net Fri May 22 05:37:19 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 22 May 2009 14:37:19 +0200 Subject: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy In-Reply-To: <4468a6770905220322q13609513k65ad14c7acfced6c@mail.gmail.com> References: <9389DD3DDD6B9144B147CE564C6599B9041D86B8BD@INBANSXCHMBSA3.in.alcatel-lucent.com> <4468a6770905220322q13609513k65ad14c7acfced6c@mail.gmail.com> Message-ID: <4A169C7F.2080305@gmx.net> This is also interesting for me, as I love freeswitch, and maintaining a single platform is easier, than handling various different ones. In the past years I did a couple of projects with OpenSER /openSIPS. These projects comprised: * registrar for the SIP user agents * handle invite messages (+ ringing, bye, ok, etc also of course) between registered user agents and user agents at external domains * rtp payload was a bit different from usual VoIP traffic (video parts, application sharing, file downloads etc.), but SDP was fine according to RFC, and OpenSER mediaproxy worked also * handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS) * The number of messages to handle was not that much (some thousand subs). For my understanding this should also be possible with Freeswitch with bypass_media. Right? Best regards Peter Ognjen Seslija schrieb: > Hello, > > FS by design is B2BUA, and it cannot route INVITEs and other SIP > methods. It can however, bridge a-leg to b-leg with or w/o media and > doing plenty other cool stuff much better than commercial projects. I > suggest joining us on irc to detail your setup so we can help you. > > Regards, > Ognjen (sekil on #freeswitch). > > On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar) > > wrote: > > Hi all, > > I want to use FreeSWITCH as a SIP transparent proxy in session > border controller application. Please let me know the changes in > configuration files required to achieve this behaviour > > Thanks very much for the help. > > Regards, > Sridhar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri May 22 05:47:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 22 May 2009 07:47:29 -0500 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> Message-ID: <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> 1) update to lastest trunk (you are at least 1000 revisions behind) 2) disable the presence debug in sofia.conf 3) enable sip trace instead "sofia profile internal siptrace on" 4) reproduce your problem. Make sure you include more of the log from before the hangup happened. The one you posted here is missing some of the info from the few seconds prior but with the incomplete info it looks like the other side sent a BYE ending the call. On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: > Thanks Brian! To answer your questions: > Freeswitch svn revision: 12148 > Centos rev: 2.6.18-92.el5 > > And apologies, actually I guess we're using g711 not 729. > > Jason: I agree it would seem to be on the switch/telco side. And, the > telco says many other people are in the same set-up as us and don't have any > issues, so they're insisting it's on our end. > > On Thu, May 21, 2009 at 7:28 PM, Brian West wrote: > >> >> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >> >> We're running FreeSwitch as part of a teleconferencing service, inside a >> telcom (so no >> internet latency/NAT issues) and using g.729 >> >> >> So you're using g729 with conferences? >> >> We are receiving some complaints of dropped calls, >> including from landlines. This means they join the conference, and x >> minutes in they simply drop. >> >> I know that cellphones tend to drop calls frequently, but landlines >> are pretty reliable, and we're hearing it a lot. From the FreeSwitch side >> of things, it just >> looks like those callers hung up (but then dialed back in just a moment >> later). >> >> I'm attaching two different snippets of the FS log files where these >> issues are occurring. >> >> >> Next time please call them .txt because you cause extra work to have to >> open them otherwise. >> >> Does anyone have any recommendations about how to troubleshoot this? >> >> Any known issues/patches in FS that could be biting us? >> >> >> Depends you failed to include some very valid info such as what version or >> svn rev you're running and what linux distro. >> >> Is there some SIP logging we can do to debug? >> >> >> Yes covered on the wiki. >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >> >> Are there any paid contractors avail who would have the expertise to look >> into this? >> >> >> email consulting at freeswitch.org >> >> Any help appreciated ... this is a major issue for us! >> >> Thanks much, >> >> -Dale >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/4b0512b7/attachment-0001.html From brian at freeswitch.org Fri May 22 06:57:20 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 22 May 2009 08:57:20 -0500 Subject: [Freeswitch-users] Retrieving voicemail In-Reply-To: <13ca621c0905220031y76944fa2re3656d2c030e0577@mail.gmail.com> References: <001801c9d972$6272ce40$27586ac0$@com> <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> <13ca621c0905220031y76944fa2re3656d2c030e0577@mail.gmail.com> Message-ID: You have old configs with the old method in it.... if you had a previous install it won't overwrite the configs with new ones. /b On May 22, 2009, at 2:31 AM, Gavin Henry wrote: > What version is that on? > > I just followed > http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and registered > with 1000/1234 and then dialed 1000 which put me straight into > voicemail. That link installs trunk, so should be the latest version > and default config? > > Thanks. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/10500da7/attachment.html From brian at freeswitch.org Fri May 22 06:59:06 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 22 May 2009 08:59:06 -0500 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: Message-ID: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> On May 22, 2009, at 12:47 AM, Jim Burke wrote: > Hey Brian, > > Will have a look at ZRTP :) > > Not sure I understand your comments regarding its all over once > receiving the 415 from the B party. Is'nt that what parm > continue_on_fail does? The fact that it sends the invite back out > sorta proves this. The A-LEG has to hangup to re-enable SRTP it can't do it if it didn't invite with it in the first place. > > The other point of interest here is that if you set application="export" data="sip_secure_media=true"/> before the first > bridge function it will include the security descriptions in the B leg > INVITE even when the A leg does not have them and the call will > succeed. The B Eyebeam will show the locked padlock while A does not. Make sure you do not answer the call before you do it. > >> From what I can see in code it is this guy that must stop it all from > happening. TFLAG_SECURE But I dont understand why :( Again you have to invite to FS with crypto it can't magically cause crypto to work unless you initiate it with your first invite. > > Regards, > Jim Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/14b08962/attachment.html From gavin.henry at gmail.com Fri May 22 07:02:53 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Fri, 22 May 2009 15:02:53 +0100 Subject: [Freeswitch-users] Retrieving voicemail In-Reply-To: References: <001801c9d972$6272ce40$27586ac0$@com> <87f2f3b90905201204s4ea4f3eej8ed9d0df8907718b@mail.gmail.com> <13ca621c0905220031y76944fa2re3656d2c030e0577@mail.gmail.com> Message-ID: <13ca621c0905220702j444d4ed5q6015e9518e537080@mail.gmail.com> 2009/5/22 Brian West : > You have old configs with the old method in it.... if you had a previous > install it won't overwrite the configs with new ones. Ah, will re-install From mszlazak at aol.com Fri May 22 07:58:18 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 22 May 2009 10:58:18 -0400 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required In-Reply-To: References: Message-ID: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com> What problems will a Windows user have when updating with Tortoise SVN? -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Thu, 21 May 2009 8:54 pm Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required FYI, We just want to let everyone know that we have made a few updates that will require a rebootstrap. One of the key updates was a security fix for libsndfile. In this particular case it won't be possible simply to "make current" like you normally do. Here is a common set of commands for a typical Linux rebootstrap: cd /usr/src/freeswitch.trunk make clean svn up ./bootstrap.sh ./configure make install NOTE: if you've got the libzrtp file and you've already run the buildzrtp.sh script then be sure to use "./configure --enable-zrtp" in the above operation. Thank you for your continued support of the FreeSWITCH project! -Michael S Collins http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/8e4739ee/attachment.html From msc at freeswitch.org Fri May 22 08:36:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 22 May 2009 08:36:59 -0700 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required In-Reply-To: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com> References: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com> Message-ID: <87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com> On Fri, May 22, 2009 at 7:58 AM, wrote: > What problems will a Windows user have when updating with Tortoise SVN? > I haven't had a chance to test it out but what I would do is update and then "rebuild solution" and see how it goes. Let us know if you run into any issues. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/aee04eb3/attachment.html From brian at freeswitch.org Fri May 22 08:42:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 22 May 2009 10:42:09 -0500 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required In-Reply-To: <87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com> References: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com> <87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com> Message-ID: <3B0C7B39-0025-4F48-8754-6DDEC5CC22B7@freeswitch.org> Windows should have to clean the solution and rebuild /b On May 22, 2009, at 10:36 AM, Michael Collins wrote: > > > On Fri, May 22, 2009 at 7:58 AM, wrote: > What problems will a Windows user have when updating with Tortoise > SVN? > > I haven't had a chance to test it out but what I would do is update > and then "rebuild solution" and see how it goes. Let us know if you > run into any issues. > > -MC Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/ed271fab/attachment-0001.html From sprice at gmail.com Fri May 22 08:45:26 2009 From: sprice at gmail.com (SP) Date: Fri, 22 May 2009 10:45:26 -0500 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> Message-ID: <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> Dasbus On Thu, May 21, 2009 at 23:26, Diego Viola wrote: > Hey guys, > > I'm about to start my own ITSP with FreeSWITCH, and I'm looking some > cool names for my VoIP company, if you know some please tell me :) > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From edpimentl at gmail.com Fri May 22 08:51:06 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 22 May 2009 11:51:06 -0400 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> Message-ID: <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> VoiceCLOUD CLOUDvoice GlobalVoice VoiceUP voicEVERYthing VoicEnterprise S/IP (Services over IP) GlobalSIP VoiPLATFORM Best regards, -E Gpro.ws -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/223c080f/attachment.html From brian at freeswitch.org Fri May 22 08:52:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 22 May 2009 10:52:00 -0500 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> Message-ID: <8FB4FDBF-05E0-455C-A41B-7AA01EEEB898@freeswitch.org> I say bkw_ On May 22, 2009, at 10:45 AM, SP wrote: > Dasbus Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/47310482/attachment.html From mszlazak at aol.com Fri May 22 09:03:30 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 22 May 2009 12:03:30 -0400 Subject: [Freeswitch-users] text to speech IVRs and MOH In-Reply-To: References: <20090519093119.2ad02225396a31c9de30536f2e338977.34d99de7f9.wbe@email04.secureserver.net><4A12ECEB.1050207@gmx.net> Message-ID: <8CBA90DFBD2E4DE-120-2C04@WEBMAIL-DZ24.sysops.aol.com> Neospeech does have the best voices. I looked at Neospeech months ago and talked to a rep. Then he quoted me a price per port of over $1000.00. Looks like they really have done some big price adjustments. -----Original Message----- From: Saeed Ahmad To: freeswitch-users at lists.freeswitch.org Sent: Fri, 22 May 2009 2:01 am Subject: Re: [Freeswitch-users] text to speech IVRs and MOH Thanks guys for a detailed reply specially pete. On Tue, May 19, 2009 at 7:31 PM, Peter P GMX wrote: Thanks for this overwiev. One question: How does this compare to Cepstral TTS? Best regards Peter pete at privateconnect.com schrieb: > I've spent the last 2-3 months on researching TTS and ASR for FS for a > project. ?Best TTS depends on what you consider important. ?Also, how > do you plan on using it. > > Here's some of the TTS engines I've run across with some pros/cons: > > Festivate Lite (flite) > Pros: > - Free (comes with FS) > - simple to use > - 16K voice sounds decent > - Completely customizable > Cons: > - 8K voice sounds horrible over cell phone > > NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl voice) > Pros: > - My selection for best soundig voices > - Recently select by Stephen Hawkings for his voice (geek points!) > - Lots of Languages supported > - Free trial available > Cons: > - Custom C-Based API (FS interface coming soon) > - Large file size (Engine + SDK + 1 Voice = 900MB) > - Support is lacking (Company beed in Korean, time zone issues, etc) > > Nuance ($500/port for 1 voice) > Pros: > - Wide Variety of Products > - Support MRCP > - Supports ASR as well (add'l fees) > - Excellent support > - Free trial > - Decent sounding voices at 8K and 16K > - Wide range of tuning parameters > Cons: > - Pricey > - Limited voice selection > - Limited support for 64-bit linux > > AT&T (NaturalVoice) (no pricing info available) > Pros: > - Big company (solid in marketplace) > - Good suppport (user and developer) > - ASP model means no software to maintain > Cons: > - ASP model incurs delay > - Voices sound too digitized > - Limited support for 64-bit linux > > Loquendo ($500/port for 1 voice + 15% addl voice) > Pros: > - Good sounding voices (almost as good as NeoSpeech) > - Wide variety of languages > - Excellent support > - Has free 30 day trial > - Supports MRCP > - Support ASR and Voice Recognition as well. (add'l fees) > - Small footprint (< 150MB) > Cons: > - Pricey > - Complicated install process > - Limited management/tuning capabilities > > In the end, it was down to NeoSpeech or Loquendo for our application. > ?We are currently running tests with NeoSpeech and assuming all goes > well, we will select them. ?Though don't let that color your opinion > too much after several "focus groups" we discovered the most important > element in the equation is does your customer/boss like the sound of > the voices, and that is a completely subjective decision. > > > -pete > > ? ? -------- Original Message -------- > ? ? Subject: [Freeswitch-users] text to speech IVRs and MOH > ? ? From: Saeed Ahmad > ? ? Date: Tue, May 19, 2009 12:40 am > ? ? To: freeswitch-users at lists.freeswitch.org > > ? ? Hi all, > > ? ? Could you guys recommend me any online text to speech IVR software > ? ? which works OK with FS. i am using AT&T site and for some IVRs i > ? ? get sample rate errors. Also some resource to download more MOH > ? ? wav files. > > ? ? Many thanks > ? ? ------------------------------------------------------------------------ > ? ? _______________________________________________ > ? ? Freeswitch-users mailing list > ? ? Freeswitch-users at lists.freeswitch.org > ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > ? ? http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/10393536/attachment.html From pete at privateconnect.com Fri May 22 11:11:48 2009 From: pete at privateconnect.com (pete at privateconnect.com) Date: Fri, 22 May 2009 11:11:48 -0700 Subject: [Freeswitch-users] text to speech IVRs and MOH Message-ID: <20090522111148.2ad02225396a31c9de30536f2e338977.1b253a8f68.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/7d8ce98c/attachment-0001.html From gcd at i.ph Fri May 22 16:56:10 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 23 May 2009 07:56:10 +0800 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <8FB4FDBF-05E0-455C-A41B-7AA01EEEB898@freeswitch.org> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <8FB4FDBF-05E0-455C-A41B-7AA01EEEB898@freeswitch.org> Message-ID: <7d0bfd8c0905221656x32912b43q63c928a0703bfccc@mail.gmail.com> how about InterTalk or InterMedia? -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Fri, May 22, 2009 at 11:52 PM, Brian West wrote: > I say bkw_ > On May 22, 2009, at 10:45 AM, SP wrote: > > Dasbus > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/8a63b1ce/attachment.html From dujinfang at gmail.com Fri May 22 17:10:02 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 23 May 2009 08:10:02 +0800 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> Message-ID: <258A7893-79B1-43B6-B798-651CAAA9CD5C@gmail.com> Voila itself is a good name. ;) VoilaVoIP On May 22, 2009, at 12:26 PM, Diego Viola wrote: > Hey guys, > > I'm about to start my own ITSP with FreeSWITCH, and I'm looking some > cool names for my VoIP company, if you know some please tell me :) > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri May 22 17:13:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 22 May 2009 19:13:09 -0500 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <258A7893-79B1-43B6-B798-651CAAA9CD5C@gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <258A7893-79B1-43B6-B798-651CAAA9CD5C@gmail.com> Message-ID: <2D234EAF-2302-48BD-BDA8-0873A06C7976@freeswitch.org> How about larynxvoip /b On May 22, 2009, at 7:10 PM, dujinfang wrote: > Voila itself is a good name. ;) > > VoilaVoIP Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/d0e70421/attachment.html From juanbackson at gmail.com Sat May 23 00:11:54 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sat, 23 May 2009 15:11:54 +0800 Subject: [Freeswitch-users] need hear solving a noise problem Message-ID: <27c25bc40905230011ha55504u3b867cad5f95d962@mail.gmail.com> Hi, I am getting problem when one UA is xlite and another UA is another sip application. When I call from xlite to a sip application, I am getting noise: I have tried these: show channels give me the following: c5f42dec-646a-4675-af40-c4d173c8a7c7,inbound,2009-05-23 10:36:30,1243089390,sofia/internal/1000 at 192.168.1.191 ,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,sofia/ 192.168.1.191/4540,XML,public,GSM,8000,GSM,8000 790d9b2a-88b9-4521-8934-31b059e04e7b,outbound,2009-05-23 10:36:30,1243089390,sofia/internal/4540,CS_CONSUME_MEDIA,1000,1000,192.168.1.193,4540,,,XML,public,,,, The sip application and xlite is working fine ( voice is clear ) if I use Asterisk with the following line in sip.conf: [4540] canreinvite=no type=friend context=sip-external allow=gsm host=dynamic [1000] canreinvite=no type=friend context=sip-external allow=gsm host=dynamic Does anyone know how to mimic the same behavior in Freeswitch? Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/4578466c/attachment.html From juanbackson at gmail.com Sat May 23 00:22:01 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sat, 23 May 2009 15:22:01 +0800 Subject: [Freeswitch-users] need hear solving a noise problem In-Reply-To: <27c25bc40905230011ha55504u3b867cad5f95d962@mail.gmail.com> References: <27c25bc40905230011ha55504u3b867cad5f95d962@mail.gmail.com> Message-ID: <27c25bc40905230022p7c5452bej9281a9a3f4300eff@mail.gmail.com> Hi, Just to follow up with this problem. If I set both xlite and sip application to use PCMU, I am still getting noise even channels show the same codec: API CALL [show(channels)] output: uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 0816684d-7a29-4814-93e4-104ffc2ed984,inbound,2009-05-23 11:25:30,1243092330,sofia/internal/1000 at 192.168.1.191 ,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,{absolute_codec_string='GSM,PCMU'}sofia/ 192.168.1.191/454044009539026,XML,public,PCMU,8000,PCMU,8000 a6f1e90c-f6a9-4ac1-9f26-fe08c5c0dd74,outbound,2009-05-23 11:25:30,1243092330,sofia/internal/454044009539026,CS_EXCHANGE_MEDIA,1000,1000,192.168.1.193,454044009539026,,,XML,public,PCMU,8000,PCMU,8000 Thanks for any suggestion. Thanks, JB On Sat, May 23, 2009 at 3:11 PM, Juan Backson wrote: > Hi, > > I am getting problem when one UA is xlite and another UA is another > sip application. > > When I call from xlite to a sip application, I am getting noise: > > I have tried these: > > > data="{absolute_codec_string='GSM,PCMU'}sofia/192.168.1.191/4540"/> > > > > > > > > > show channels give me the following: > > c5f42dec-646a-4675-af40-c4d173c8a7c7,inbound,2009-05-23 > 10:36:30,1243089390,sofia/internal/1000 at 192.168.1.191 > ,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,sofia/ > 192.168.1.191/4540,XML,public,GSM,8000,GSM,8000 > 790d9b2a-88b9-4521-8934-31b059e04e7b,outbound,2009-05-23 > 10:36:30,1243089390,sofia/internal/4540,CS_CONSUME_MEDIA,1000,1000,192.168.1.193,4540,,,XML,public,,,, > > The sip application and xlite is working fine ( voice is clear ) if I use > Asterisk with the following line in sip.conf: > > [4540] > canreinvite=no > type=friend > context=sip-external > allow=gsm > host=dynamic > > [1000] > canreinvite=no > type=friend > context=sip-external > allow=gsm > host=dynamic > > > Does anyone know how to mimic the same behavior in Freeswitch? > > Thanks, > JB > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/e8fccc10/attachment.html From oseslija at gmail.com Sat May 23 00:41:21 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Sat, 23 May 2009 09:41:21 +0200 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> Message-ID: <4468a6770905230041m3134a6fah27a2bce56dfcaefa@mail.gmail.com> I vote for viotel. Regards, Ognjen On Fri, May 22, 2009 at 6:26 AM, Diego Viola wrote: > Hey guys, > > I'm about to start my own ITSP with FreeSWITCH, and I'm looking some > cool names for my VoIP company, if you know some please tell me :) > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/8a6d9871/attachment-0001.html From oseslija at gmail.com Sat May 23 00:48:24 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Sat, 23 May 2009 09:48:24 +0200 Subject: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy In-Reply-To: <4A169C7F.2080305@gmx.net> References: <9389DD3DDD6B9144B147CE564C6599B9041D86B8BD@INBANSXCHMBSA3.in.alcatel-lucent.com> <4468a6770905220322q13609513k65ad14c7acfced6c@mail.gmail.com> <4A169C7F.2080305@gmx.net> Message-ID: <4468a6770905230048v552b6174q6b4c7beab97a2a83@mail.gmail.com> It can be done with or without media. FS does not relay SIP messages transparently as OpenSER, but act as a SIP UA to each call leg. This gives an oportunity to place rtp in between streams, do call recording and all other softswitch stuff. I use FreeSWITCH as hosted PBX solution, basically hosting multiple SIP domains each of them representing a separate PBX for a different company with separate dialplans etc. Ognjen On Fri, May 22, 2009 at 2:37 PM, Peter P GMX wrote: > This is also interesting for me, as I love freeswitch, and maintaining a > single platform is easier, than handling various different ones. > In the past years I did a couple of projects with OpenSER /openSIPS. > These projects comprised: > > * registrar for the SIP user agents > * handle invite messages (+ ringing, bye, ok, etc also of course) > between registered user agents and user agents at external domains > * rtp payload was a bit different from usual VoIP traffic (video > parts, application sharing, file downloads etc.), but SDP was fine > according to RFC, and OpenSER mediaproxy worked also > * handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS) > * The number of messages to handle was not that much (some thousand > subs). > > For my understanding this should also be possible with Freeswitch with > bypass_media. Right? > > Best regards > Peter > > > > Ognjen Seslija schrieb: > > Hello, > > > > FS by design is B2BUA, and it cannot route INVITEs and other SIP > > methods. It can however, bridge a-leg to b-leg with or w/o media and > > doing plenty other cool stuff much better than commercial projects. I > > suggest joining us on irc to detail your setup so we can help you. > > > > Regards, > > Ognjen (sekil on #freeswitch). > > > > On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar) > > > > wrote: > > > > Hi all, > > > > I want to use FreeSWITCH as a SIP transparent proxy in session > > border controller application. Please let me know the changes in > > configuration files required to achieve this behaviour > > > > Thanks very much for the help. > > > > Regards, > > Sridhar > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/2054399e/attachment.html From asannucci at gmail.com Sat May 23 05:24:59 2009 From: asannucci at gmail.com (bakko) Date: Sat, 23 May 2009 14:24:59 +0200 Subject: [Freeswitch-users] Problem with compiling Message-ID: <2058FBCAB1AA433BAEA1938F90859047@voztovoice> Hi, when i try to compile the latest trunk i receive this error: touch .version make[1]: Entering directory `/usr/src/freeswitch/libs/apr' make[1]: *** No targets specified and no makefile found. Stop. make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' make: *** [libs/apr/libapr-1.la] Error 2 What i do: ./bootstrap.sh ./configure make Thank's Regards From Prometheus001 at gmx.net Sat May 23 05:38:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 23 May 2009 14:38:25 +0200 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <4468a6770905230041m3134a6fah27a2bce56dfcaefa@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <4468a6770905230041m3134a6fah27a2bce56dfcaefa@mail.gmail.com> Message-ID: <4A17EE41.3080203@gmx.net> Just a side notice about how to name a company. If you use a "descriptive" name e.g. GlobalSIP as sugested before, it may be difficult to register this name later on as a brand name when your company becomes successful. At least here in Europe it is not possible to register a brand name when the name itself describes the business or the techique used. Thats the reason why big companies nowadays use these strange names like e.g. ABALA, which seem to not make any sense at all at a first glance. But these names can eaysily be registered as brand names. Best regards Peter Ognjen Seslija schrieb: > I vote for viotel. > > Regards, > Ognjen > On Fri, May 22, 2009 at 6:26 AM, Diego Viola > wrote: > > Hey guys, > > I'm about to start my own ITSP with FreeSWITCH, and I'm looking some > cool names for my VoIP company, if you know some please tell me :) > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From testeador01 at gmail.com Sat May 23 06:21:25 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 23 May 2009 08:21:25 -0500 Subject: [Freeswitch-users] Problem with compiling In-Reply-To: <2058FBCAB1AA433BAEA1938F90859047@voztovoice> References: <2058FBCAB1AA433BAEA1938F90859047@voztovoice> Message-ID: did you try as it says in http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/014456.html ?: cd /usr/src/freeswitch.trunk make clean svn up ./bootstrap.sh ./configure make install 2009/5/23 bakko > Hi, > > when i try to compile the latest trunk i receive this error: > > touch .version > make[1]: Entering directory `/usr/src/freeswitch/libs/apr' > make[1]: *** No targets specified and no makefile found. Stop. > make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' > make: *** [libs/apr/libapr-1.la] Error 2 > > What i do: > > ./bootstrap.sh > ./configure > make > > Thank's > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/fe7afb5c/attachment.html From steveu at coppice.org Sat May 23 07:37:17 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 23 May 2009 22:37:17 +0800 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <4A17EE41.3080203@gmx.net> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <4468a6770905230041m3134a6fah27a2bce56dfcaefa@mail.gmail.com> <4A17EE41.3080203@gmx.net> Message-ID: <4A180A1D.2080006@coppice.org> Peter P GMX wrote: > Just a side notice about how to name a company. > If you use a "descriptive" name e.g. GlobalSIP as sugested before, it > may be difficult to register this name later on as a brand name when > your company becomes successful. > At least here in Europe it is not possible to register a brand name when > the name itself describes the business or the techique used. > Thats the reason why big companies nowadays use these strange names like > There have been some high profile renamings in Europe, where the original name was too descriptive to trademark. For example, you couldn't trademark a name like "Windows" for a windowing system... oh, wait.. > e.g. ABALA, which seem to not make any sense at all at a first glance. > But these names can eaysily be registered as brand names. > Not really. The names are getting weirder as the range of available .com options shrinks. Steve From mszlazak at aol.com Sat May 23 08:33:25 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 23 May 2009 11:33:25 -0400 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required In-Reply-To: <87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com> References: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com> <87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com> Message-ID: <8CBA9D2F2785BFA-1248-1C92@webmail-db05.sysops.aol.com> I get a lot of these errors on vc++ express 2008: 59>Linking... 59>LINK : fatal error LNK1181: cannot open input file '..\..\..\..\w32\library\debug\freeswitchcore.lib' 59>Build log was saved at "file://c:\Documents and Settings\Mark Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv\Win32\Debug\BuildLog.htm" 59>mod_fsv - 1 error(s), 1 warning(s) -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Fri, 22 May 2009 8:36 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required On Fri, May 22, 2009 at 7:58 AM, wrote: What problems will a Windows user have when updating with Tortoise SVN? I haven't had a chance to test it out but what I would do is update and then "rebuild solution" and see how it goes. Let us know if you run into any issues. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/6f7196c2/attachment-0001.html From msc at freeswitch.org Sat May 23 09:42:30 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 23 May 2009 09:42:30 -0700 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required In-Reply-To: <8CBA9D2F2785BFA-1248-1C92@webmail-db05.sysops.aol.com> References: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com> <87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com> <8CBA9D2F2785BFA-1248-1C92@webmail-db05.sysops.aol.com> Message-ID: Did you try clean solution as brian suggested? -MC Sent from my iPhone On May 23, 2009, at 8:33 AM, mszlazak at aol.com wrote: > I get a lot of these errors on vc++ express 2008: > > > 59>Linking... > 59>LINK : fatal error LNK1181: cannot open input file '..\..\..\.. > \w32\library\debug\freeswitchcore.lib' > 59>Build log was saved at "file://c:\Documents and Settings\Mark > Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv > \Win32\Debug\BuildLog.htm" > 59>mod_fsv - 1 error(s), 1 warning(s) > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, 22 May 2009 8:36 am > Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN > Updates - Bootstrap Required > > > > On Fri, May 22, 2009 at 7:58 AM, wrote: > What problems will a Windows user have when updating with Tortoise > SVN? > > I haven't had a chance to test it out but what I would do is update > and then "rebuild solution" and see how it goes. Let us know if you > run into any issues. > > -MC > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > We found the real 'Hotel California' and the 'Seinfeld' diner. What > will you find? Explore WhereItsAt.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/9c666d3e/attachment.html From mszlazak at aol.com Sat May 23 12:48:07 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 23 May 2009 15:48:07 -0400 Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required In-Reply-To: References: <8CBA904E00C8688-120-2859@WEBMAIL-DZ24.sysops.aol.com><87f2f3b90905220836k46f3bf18wa0f6b78a97a640ee@mail.gmail.com><8CBA9D2F2785BFA-1248-1C92@webmail-db05.sysops.aol.com> Message-ID: <8CBA9F687AFC61C-FC4-199D@WEBMAIL-DF12.sysops.aol.com> Yes I did. -----Original Message----- From: Michael S Collins To: freeswitch-users at lists.freeswitch.org Sent: Sat, 23 May 2009 9:42 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required Did you try clean solution as brian suggested? -MC Sent from my iPhone On May 23, 2009, at 8:33 AM, mszlazak at aol.com wrote: I get a lot of these errors on vc++ express 2008: 59>Linking... 59>LINK : fatal error LNK1181: cannot open input file '..\..\..\..\w32\library\debug\freeswitchcore.lib' 59>Build log was saved at "file://c:\Documents and Settings\Mark Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv\Win32\Debug\BuildLog.htm" 59>mod_fsv - 1 error(s), 1 warning(s) -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Fri, 22 May 2009 8:36 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required On Fri, May 22, 2009 at 7:58 AM, wrote: What problems will a Windows user have when updating with Tortoise SVN? I haven't had a chance to test it out but what I would do is update and then "rebuild solution" and see how it goes. Let us know if you run into any issues. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org We found the real 'Hotel California' and the 'Seinfeld' diner. What will you find? Explore WhereItsAt.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/d9315b8f/attachment.html From d at unwire.it Fri May 22 10:57:50 2009 From: d at unwire.it (Darin Weeks) Date: Fri, 22 May 2009 10:57:50 -0700 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> Message-ID: <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> I just went through naming my own thing -- pretty happy with how it came out. I'm setting up a wisp for my 2 square mile city in los angeles metro area. The name: UNWIRE.IT http://unwire.it :-) The other one I was/am toying with is WEHO.FI (short for west hollywood) On Fri, May 22, 2009 at 8:51 AM, EdPimentl wrote: > VoiceCLOUD > CLOUDvoice > GlobalVoice > VoiceUP > voicEVERYthing > VoicEnterprise > S/IP (Services over IP) > GlobalSIP > VoiPLATFORM > > > Best regards, > -E > Gpro.ws > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/c483efa1/attachment.html From simon at imaginator.com Sat May 23 11:23:15 2009 From: simon at imaginator.com (Simon Tennant) Date: Sat, 23 May 2009 20:23:15 +0200 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? Message-ID: <4A183F13.2030108@imaginator.com> I have a pretty basic setup where many nokia sip clients (user agent: Nokia-SIPUA) behind different home NATs need to connect to a publicly accessible IP address running Freeswitch (freeswitch is then connected to SIP trunks and either routes the calls between handsets or outbound to the PSTN). I'm wondering if anyone could share their configurations for getting this to work *well*. I never really got it working well enough to be considered production quality on Asterisk but was hoping that someone had gone through some learning that would save me some debugging hours. Ideally I'm looking for a combination of client configuration options and server configuration that works well together - like should I use the IP address of the server on the client or a DNS name? And whats the best way to configure Freeswitch to make like easier for a random sip-capable Nokia handset to connect to freeswitch. Could anyone post working freeswitch configurations before I start on what turned out to be an endless and ultimately unsuccessful pursuit in Asterisk-land? S. -- Simon Tennant _____________________________________________ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: simon at buddycloud.com From diego.viola at gmail.com Sat May 23 13:31:25 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 23 May 2009 16:31:25 -0400 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> Message-ID: <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> Thanks everyone, we just choose our own name :). On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote: > > I just went through naming my own thing -- pretty happy with how it came > out. ?I'm setting up a wisp for my 2 square mile city in los angeles metro > area. > The name: ?UNWIRE.IT > http://unwire.it > :-) > The other one I was/am toying with is WEHO.FI > (short for west hollywood) > On Fri, May 22, 2009 at 8:51 AM, EdPimentl wrote: >> >> VoiceCLOUD >> CLOUDvoice >> GlobalVoice >> VoiceUP >> voicEVERYthing >> VoicEnterprise >> S/IP (Services over IP) >> GlobalSIP >> VoiPLATFORM >> >> >> Best regards, >> -E >> Gpro.ws >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > ? ?d at unwire.it ? ? ? ?http://unwire.it > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sat May 23 14:30:54 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 23 May 2009 16:30:54 -0500 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> Message-ID: I vote for Moose Penis Telecom /b On May 23, 2009, at 3:31 PM, Diego Viola wrote: > Thanks everyone, we just choose our own name :). > > On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote: >> >> I just went through naming my own thing -- pretty happy with how it >> came >> out. I'm setting up a wisp for my 2 square mile city in los >> angeles metro >> area. >> The name: UNWIRE.IT >> http://unwire.it >> :-) >> The other one I was/am toying with is WEHO.FI >> (short for west hollywood) >> On Fri, May 22, 2009 at 8:51 AM, EdPimentl >> wrote: >>> >>> VoiceCLOUD >>> CLOUDvoice >>> GlobalVoice >>> VoiceUP >>> voicEVERYthing >>> VoicEnterprise >>> S/IP (Services over IP) >>> GlobalSIP >>> VoiPLATFORM >>> >>> >>> Best regards, >>> -E >>> Gpro.ws >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( >> broadband internet for west hollywood >> d at unwire.it http://unwire.it >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From evilla at chipoly.com Sat May 23 15:08:43 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Sat, 23 May 2009 16:08:43 -0600 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> Message-ID: <005301c9dbf3$09a1e2f0$1ce5a8d0$@com> Hahaha that was funny Brian!!! Ing. Edwin Villarreal World Net Commerce SA CV Tel. (667) 715-6003 Cel. (667) 751-6186 www.clubwnc.com www.wncmx.com -----Mensaje original----- De: Brian West [mailto:brian at freeswitch.org] Enviado el: s?bado, 23 de mayo de 2009 03:31 p.m. Para: freeswitch-users at lists.freeswitch.org Asunto: Re: [Freeswitch-users] Cool names for my VoIP company I vote for Moose Penis Telecom /b On May 23, 2009, at 3:31 PM, Diego Viola wrote: > Thanks everyone, we just choose our own name :). > > On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote: >> >> I just went through naming my own thing -- pretty happy with how it >> came >> out. I'm setting up a wisp for my 2 square mile city in los >> angeles metro >> area. >> The name: UNWIRE.IT >> http://unwire.it >> :-) >> The other one I was/am toying with is WEHO.FI >> (short for west hollywood) >> On Fri, May 22, 2009 at 8:51 AM, EdPimentl >> wrote: >>> >>> VoiceCLOUD >>> CLOUDvoice >>> GlobalVoice >>> VoiceUP >>> voicEVERYthing >>> VoicEnterprise >>> S/IP (Services over IP) >>> GlobalSIP >>> VoiPLATFORM >>> >>> >>> Best regards, >>> -E >>> Gpro.ws >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( >> broadband internet for west hollywood >> d at unwire.it http://unwire.it >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From fvillarroel at yahoo.com Sat May 23 20:35:07 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Sat, 23 May 2009 20:35:07 -0700 (PDT) Subject: [Freeswitch-users] FS & Wikipbx Help Message-ID: <557185.81756.qm@web34302.mail.mud.yahoo.com> Dear All, I am new user of Freeswitch and searching information i find wikipbx: http://www.wikipbx.org/ My install of wikipbx was succesfully, but i have problem for registering a Softphone Xlite for testing; i look the next warning in FS_CLI: freeswitch at internal> 2009-04-28 06:13:58 [WARNING] sofia_reg.c:1701 sofia_reg_parse_auth() Can't find user [223 at 192.168.1.150] You must define a domain called '192.168.1.150' in your directory and add a user with the id="223" attribute and you must configure your device to use the proper domain in it's authentication credentials. When i solve this problem i want make a gateway between FS and Asterisk, for sending traffic from FS to Asterisk. If any could help me to resolve register ATA or Softphones with wikipbx. Best regards, Fernando From asannucci at gmail.com Sat May 23 22:47:57 2009 From: asannucci at gmail.com (bakko) Date: Sun, 24 May 2009 07:47:57 +0200 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: <4A183F13.2030108@imaginator.com> References: <4A183F13.2030108@imaginator.com> Message-ID: <30153D4481AF4017B2A9DDE90320905B@voztovoice> I have used a Nokia 6300i (SIP client) with asterisk without problems. With FreeSWITCH i have some problem (i think NAT problem) but i can make and receive calls. When i will have time i will write a post in the wiki. Bye From simon at imaginator.com Sun May 24 01:07:09 2009 From: simon at imaginator.com (Simon Tennant) Date: Sun, 24 May 2009 10:07:09 +0200 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: <30153D4481AF4017B2A9DDE90320905B@voztovoice> References: <4A183F13.2030108@imaginator.com> <30153D4481AF4017B2A9DDE90320905B@voztovoice> Message-ID: <4A19002D.1080800@imaginator.com> I would really appreciate this. Some of the problems I have been unable to solve are: * inbound callerid presentation. SIP inbound to nokia have "0123456" (quotes included) instead of just a number that would then be matched with a phone book entry to display the name of the caller. * random call drop-outs (am debugging on the wifi ap at the moment) * intermittent NAT one-way audio problems S. bakko wrote: > I have used a Nokia 6300i (SIP client) with asterisk without problems. > > With FreeSWITCH i have some problem (i think NAT problem) but i can make and > receive calls. > > When i will have time i will write a post in the wiki. > > Bye > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Simon Tennant _____________________________________________ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: simon at buddycloud.com From asannucci at gmail.com Sun May 24 07:17:42 2009 From: asannucci at gmail.com (bakko) Date: Sun, 24 May 2009 16:17:42 +0200 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: <4A19002D.1080800@imaginator.com> References: <4A183F13.2030108@imaginator.com><30153D4481AF4017B2A9DDE90320905B@voztovoice> <4A19002D.1080800@imaginator.com> Message-ID: <3CE3091789184A0AB909C026F9E8D8CA@voztovoice> I just installing FS in my new server. When i can make tests y will write you. Regards From david.villasmil.work at gmail.com Sun May 24 08:47:40 2009 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 24 May 2009 17:47:40 +0200 Subject: [Freeswitch-users] FS & Wikipbx Help In-Reply-To: <557185.81756.qm@web34302.mail.mud.yahoo.com> References: <557185.81756.qm@web34302.mail.mud.yahoo.com> Message-ID: <9853f4ff0905240847m6dcd4549seb959fa95ccf2c7c@mail.gmail.com> Hello Fernando, The best you can use is this one: http://wiki.freeswitch.org/wiki/Getting_Started_Guide very easy ;) David On Sun, May 24, 2009 at 5:35 AM, FERNANDO VILLARROEL wrote: > > Dear All, > > I am new user of Freeswitch and searching information i find wikipbx: > > http://www.wikipbx.org/ > > My install of wikipbx was succesfully, but i have problem for registering a > Softphone Xlite for testing; i look the next warning in FS_CLI: > > freeswitch at internal> 2009-04-28 06:13:58 [WARNING] sofia_reg.c:1701 > sofia_reg_parse_auth() Can't find user [223 at 192.168.1.150] > You must define a domain called '192.168.1.150' in your directory and add a > user with the id="223" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > When i solve this problem i want make a gateway between FS and Asterisk, > for sending traffic from FS to Asterisk. > > If any could help me to resolve register ATA or Softphones with wikipbx. > > Best regards, > > Fernando > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090524/cc2f4977/attachment.html From asimwarsi at yahoo.com Sun May 24 03:21:17 2009 From: asimwarsi at yahoo.com (asim iftikhar) Date: Sun, 24 May 2009 03:21:17 -0700 (PDT) Subject: [Freeswitch-users] freeswitch basic help Message-ID: <861258.60216.qm@web50601.mail.re2.yahoo.com> i need a manual sort of thing to help out for implementing a basic IVR using freeswitch --- On Sun, 5/24/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 35, Issue 116 To: freeswitch-users at lists.freeswitch.org Date: Sunday, May 24, 2009, 1:31 AM Send Freeswitch-users mailing list submissions to ??? freeswitch-users at 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 at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at 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. Re: IMPORTANT: Latest FreeSWITCH SVN Updates -??? Bootstrap ? ? ? Required (Michael S Collins) ???2. Re: IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap ? ? ? Required (mszlazak at aol.com) ???3. Re: Cool names for my VoIP company (Darin Weeks) ???4. Nokia SIP clients: share your working config? (Simon Tennant) ???5. Re: Cool names for my VoIP company (Diego Viola) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 09:42:30 -0700 From: Michael S? Collins Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN ??? Updates -??? Bootstrap Required To: "freeswitch-users at lists.freeswitch.org" ??? Message-ID: Content-Type: text/plain; charset="us-ascii" Did you try clean solution as brian suggested? -MC Sent from my iPhone On May 23, 2009, at 8:33 AM, mszlazak at aol.com wrote: > I get a lot of these errors on vc++ express 2008: > > > 59>Linking... > 59>LINK : fatal error LNK1181: cannot open input file '..\..\..\.. > \w32\library\debug\freeswitchcore.lib' > 59>Build log was saved at "file://c:\Documents and Settings\Mark? > Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv > \Win32\Debug\BuildLog.htm" > 59>mod_fsv - 1 error(s), 1 warning(s) > > > -----Original Message----- > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, 22 May 2009 8:36 am > Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN? > Updates - Bootstrap Required > > > > On Fri, May 22, 2009 at 7:58 AM, wrote: > What problems will a Windows user have when updating with Tortoise? > SVN? > > I haven't had a chance to test it out but what I would do is update? > and then "rebuild solution" and see how it goes. Let us know if you? > run into any issues. > > -MC > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > We found the real 'Hotel California' and the 'Seinfeld' diner. What? > will you find? Explore WhereItsAt.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/9c666d3e/attachment-0001.html ------------------------------ Message: 2 Date: Sat, 23 May 2009 15:48:07 -0400 From: mszlazak at aol.com Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN ??? Updates - Bootstrap Required To: freeswitch-users at lists.freeswitch.org Message-ID: <8CBA9F687AFC61C-FC4-199D at WEBMAIL-DF12.sysops.aol.com> Content-Type: text/plain; charset="us-ascii" Yes I did. -----Original Message----- From: Michael S? Collins To: freeswitch-users at lists.freeswitch.org Sent: Sat, 23 May 2009 9:42 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required Did you try clean solution as brian suggested? -MC Sent from my iPhone On May 23, 2009, at 8:33 AM, mszlazak at aol.com wrote: I get a lot of these errors on vc++ express 2008: 59>Linking... 59>LINK : fatal error LNK1181: cannot open input file '..\..\..\..\w32\library\debug\freeswitchcore.lib' 59>Build log was saved at "file://c:\Documents and Settings\Mark Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv\Win32\Debug\BuildLog.htm" 59>mod_fsv - 1 error(s), 1 warning(s) -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Fri, 22 May 2009 8:36 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required On Fri, May 22, 2009 at 7:58 AM,? wrote: What problems will a Windows user have when updating with Tortoise SVN? I haven't had a chance to test it out but what I would do is update and then "rebuild solution" and see how it goes. Let us know if you run into any issues. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org We found the real 'Hotel California' and the 'Seinfeld' diner. What will you find? Explore WhereItsAt.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/d9315b8f/attachment-0001.html ------------------------------ Message: 3 Date: Fri, 22 May 2009 10:57:50 -0700 From: Darin Weeks Subject: Re: [Freeswitch-users] Cool names for my VoIP company To: freeswitch-users at lists.freeswitch.org Message-ID: ??? <989132e70905221057h49f0a613pae9215eb9afa662d at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I just went through naming my own thing -- pretty happy with how it came out.? I'm setting up a wisp for my 2 square mile city in los angeles metro area. The name:? UNWIRE.IT http://unwire.it :-) The other one I was/am toying with is WEHO.FI (short for west hollywood) On Fri, May 22, 2009 at 8:51 AM, EdPimentl wrote: > VoiceCLOUD > CLOUDvoice > GlobalVoice > VoiceUP > voicEVERYthing > VoicEnterprise > S/IP (Services over IP) > GlobalSIP > VoiPLATFORM > > > Best regards, > -E > Gpro.ws > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood ???d at unwire.it? ? ? ? http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090522/c483efa1/attachment-0001.html ------------------------------ Message: 4 Date: Sat, 23 May 2009 20:23:15 +0200 From: Simon Tennant Subject: [Freeswitch-users] Nokia SIP clients: share your working ??? config? To: freeswitch-users at lists.freeswitch.org Message-ID: <4A183F13.2030108 at imaginator.com> Content-Type: text/plain; charset=ISO-8859-1 I have a pretty basic setup where many nokia sip clients (user agent: Nokia-SIPUA) behind different home NATs need to connect to a publicly accessible IP address running Freeswitch (freeswitch is then connected to SIP trunks and either routes the calls between handsets or outbound to the PSTN). I'm wondering if anyone could share their configurations for getting this to work *well*.? I never really got it working well enough to be considered production quality on Asterisk but was hoping that someone had gone through some learning that would save me some debugging hours. Ideally I'm looking for a combination of client configuration options and server configuration that works well together - like should I use the IP address of the server on the client or a DNS name?? And whats the best way to configure Freeswitch to make like easier for a random sip-capable Nokia handset to connect to freeswitch. Could anyone post working freeswitch configurations before I start on what turned out to be an endless and ultimately unsuccessful pursuit in Asterisk-land? S. -- Simon Tennant _____________________________________________ fixed: .uk +44 20 7043 6756? ? ? ? ? .de +49 89 420 955 854? ? mob: .uk +44 78 5335 6047? ? ? ? ? .de +49 17 8545 0880 xmpp: simon at buddycloud.com ------------------------------ Message: 5 Date: Sat, 23 May 2009 16:31:25 -0400 From: Diego Viola Subject: Re: [Freeswitch-users] Cool names for my VoIP company To: freeswitch-users at lists.freeswitch.org Message-ID: ??? <86a32abc0905231331i3a71c1f2gc83a215266ffc96e at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Thanks everyone, we just choose our own name :). On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote: > > I just went through naming my own thing -- pretty happy with how it came > out. ?I'm setting up a wisp for my 2 square mile city in los angeles metro > area. > The name: ?UNWIRE.IT > http://unwire.it > :-) > The other one I was/am toying with is WEHO.FI > (short for west hollywood) > On Fri, May 22, 2009 at 8:51 AM, EdPimentl wrote: >> >> VoiceCLOUD >> CLOUDvoice >> GlobalVoice >> VoiceUP >> voicEVERYthing >> VoicEnterprise >> S/IP (Services over IP) >> GlobalSIP >> VoiPLATFORM >> >> >> Best regards, >> -E >> Gpro.ws >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > ? ?d at unwire.it ? ? ? ?http://unwire.it > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 35, Issue 116 ************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090524/7ed029ae/attachment-0001.html From edpimentl at gmail.com Sun May 24 09:16:48 2009 From: edpimentl at gmail.com (EdPimentl) Date: Sun, 24 May 2009 12:16:48 -0400 Subject: [Freeswitch-users] freeswitch basic help In-Reply-To: <861258.60216.qm@web50601.mail.re2.yahoo.com> References: <861258.60216.qm@web50601.mail.re2.yahoo.com> Message-ID: <9dc4a1670905240916l7a6b148dq2470e2a938fb28c0@mail.gmail.com> Here is a start: http://wiki.freeswitch.org/wiki/IVR http://docs.freeswitch.org/group__switch__ivr__menu.html http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate http://wiki.freeswitch.org/wiki/Mod_commands Best regards, -E Gpro.ws -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090524/d45d136f/attachment.html From niall.crosby at gmail.com Sun May 24 14:09:53 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 24 May 2009 22:09:53 +0100 Subject: [Freeswitch-users] Sangoma configuration Message-ID: <4aec92830905241409k48cd7430uc15516759c561989@mail.gmail.com> Hi List, Am about to buy a Sangoma A101 Single Port T1/E1/J1 to plug our Freeswitch box into an E1 at the telco. Is there any reference on how to set up Freeswitch to interface with this? thanks, Niall. -- -- Sremium Ltd. Reg Number: 451937 Mobile: +353 (0)87 2393174 Web: www.sremium.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of Sremium. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. From jason at jasonjgw.net Sun May 24 18:17:36 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 25 May 2009 11:17:36 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module Message-ID: <20090525011736.GA25198@jdc.jasonjgw.net> I've already discussed this with a few members of the community, but I would like to raise it with a wider FreeSWITCH audience. Since upgrading to libtool 2.2.6a (now the default in debian testing), I can't successfully link mod_portaudio.so. The system is Debian Testing, x86_64 architecture. ldd -d mod_portaudio.so returns the following (after listing the library dependencies): undefined symbol: snd_config (./mod_portaudio.so) snd_config is from the Alsa library, libasound. mod_portaudio.la shows a reference to /usr/lib/libasound.la, all of which looks correct. I have also examined the build logs, without finding the problem, as there are no errors reported during linking and I didn't find any linker invocations with library parameters apparently missing. The ldd -d output: undefined symbol: snd_config (./mod_portaudio.so) linux-vdso.so.1 => (0x00007fffd31ff000) libm.so.6 => /lib/libm.so.6 (0x00007f0ccacf7000) libfreeswitch.so.1 => /opt/freeswitch/lib/libfreeswitch.so.1 (0x00007f0c ca8c6000) libpthread.so.0 => /lib/libpthread.so.0 (0x00007f0cca6aa000) libc.so.6 => /lib/libc.so.6 (0x00007f0cca357000) libncurses.so.5 => /lib/libncurses.so.5 (0x00007f0cca118000) libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f0cc9e08000) libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f0cc9bed000) libodbc.so.1 => /usr/lib/libodbc.so.1 (0x00007f0cc998d000) /lib64/ld-linux-x86-64.so.2 (0x00007f0ccb1d1000) libdl.so.2 => /lib/libdl.so.2 (0x00007f0cc9788000) libltdl.so.7 => /usr/lib/libltdl.so.7 (0x00007f0cc957f000) I spent several hours yesterday trying to debug this one. Any suggestions would be welcome. At this stage, I'm not sure whether it's a FreeSWITCH problem or a bug in libtool, or elsewhere. From mashudiflexi at telkom.co.id Sun May 24 19:08:48 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Mon, 25 May 2009 09:08:48 +0700 Subject: [Freeswitch-users] Sangoma configuration In-Reply-To: <4aec92830905241409k48cd7430uc15516759c561989@mail.gmail.com> References: <4aec92830905241409k48cd7430uc15516759c561989@mail.gmail.com> Message-ID: <4A19FDB0.5090200@telkom.co.id> Hi, Niall Crosby, may it could help : http://www.freeswitch.de/xwiki/bin/viewrev/OpenZAP/WebHome?rev=8.1 I have Sangoma A104D quad E1 , I follow that link, and it's work. Niall Crosby wrote: > Hi List, > > Am about to buy a Sangoma A101 Single Port T1/E1/J1 to plug our > Freeswitch box into an E1 at the telco. > > Is there any reference on how to set up Freeswitch to interface with this? > > thanks, > Niall. > > > > ***************************************** Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. From jason at jasonjgw.net Sun May 24 18:58:04 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 25 May 2009 11:58:04 +1000 Subject: [Freeswitch-users] Linking problems with mod_portaudio.so module In-Reply-To: <20090525011736.GA25198@jdc.jasonjgw.net> References: <20090525011736.GA25198@jdc.jasonjgw.net> Message-ID: <20090525015804.GA30461@jdc.jasonjgw.net> With apologies for the incorrect address in the header, if you reply to this follow-up instead of the original message we should be fine for the remainder of the thread. From jason at jasonjgw.net Sun May 24 19:17:37 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 25 May 2009 12:17:37 +1000 Subject: [Freeswitch-users] FS & Wikipbx Help In-Reply-To: <557185.81756.qm@web34302.mail.mud.yahoo.com> References: <557185.81756.qm@web34302.mail.mud.yahoo.com> Message-ID: <20090525021737.GA31033@jdc.jasonjgw.net> FERNANDO VILLARROEL wrote: > My install of wikipbx was succesfully, but i have problem for registering a > Softphone Xlite for testing; i look the next warning in FS_CLI: > > freeswitch at internal> 2009-04-28 06:13:58 [WARNING] sofia_reg.c:1701 > sofia_reg_parse_auth() Can't find user [223 at 192.168.1.150] You must define a > domain called '192.168.1.150' in your directory and add a user with the > id="223" attribute and you must configure your device to use the proper > domain in it's authentication credentials. > > When i solve this problem i want make a gateway between FS and Asterisk, for > sending traffic from FS to Asterisk. > > If any could help me to resolve register ATA or Softphones with wikipbx. If WikiPBX has its own mailing list, I think you would receive better help there. After installing FreeSWITCH, try registering to one of the extensions (1000 to 1019) provided in the default configuration. If that works, which it should, you should then try creating your own extension, or modifying an existing one as required. The error message you quote says very explicitly what is wrong, but without access to your FreeSWITCH configuration, it is impossible to help any further, except to suggest that you start with the default configuration and work from there. From wasim at convergence.pk Sun May 24 20:05:02 2009 From: wasim at convergence.pk (Wasim Baig) Date: Mon, 25 May 2009 09:05:02 +0600 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: <3CE3091789184A0AB909C026F9E8D8CA@voztovoice> References: <4A183F13.2030108@imaginator.com> <30153D4481AF4017B2A9DDE90320905B@voztovoice> <4A19002D.1080800@imaginator.com> <3CE3091789184A0AB909C026F9E8D8CA@voztovoice> Message-ID: On Sun, May 24, 2009 at 8:17 PM, bakko wrote: > I just installing FS in my new server. When i can make tests y will write > you. bakko: its really quite simple ... and should work straight out of the box first install your server - http://wiki.freeswitch.org/wiki/Installation_Guide then get started using the - http://wiki.freeswitch.org/wiki/Getting_Started_Guide then configure Nokia - http://wiki.freeswitch.org/wiki/Nokia_N95 -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/81ad16f7/attachment.html From edpimentl at gmail.com Sun May 24 20:10:05 2009 From: edpimentl at gmail.com (EdPimentl) Date: Sun, 24 May 2009 23:10:05 -0400 Subject: [Freeswitch-users] freeswitch basic help In-Reply-To: <861258.60216.qm@web50601.mail.re2.yahoo.com> References: <861258.60216.qm@web50601.mail.re2.yahoo.com> Message-ID: <9dc4a1670905242010s1f312805uc446321454bbb144@mail.gmail.com> *Here is a better FreeSwitch IVR starter guide..* http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/IVR http://docs.freeswitch.org/group__switch__ivr__menu.html http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_phrase http://wiki.freeswitch.org/wiki/Mod_cepstral http://wiki.freeswitch.org/wiki/Session_speak http://wiki.freeswitch.org/wiki/Session http://wiki.freeswitch.org/wiki/Mod_commands http://wiki.freeswitch.org/wiki/Which_scripting_language_should_I_use%3F http://wiki.freeswitch.org/wiki/Mod_lua http://wiki.freeswitch.org/wiki/Mod_python http://wiki.freeswitch.org/wiki/Javascript * Simple js sample apps .. * http://wiki.freeswitch.org/wiki/Javascript_QuickStart http://wiki.freeswitch.org/wiki/Javascript_FAQ http://wiki.freeswitch.org/wiki/Javascript_Examples Best regards, -E Gpro.ws http://Twitter.com/edpimentl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090524/8d0460b2/attachment.html From mashudiflexi at telkom.co.id Sun May 24 19:58:54 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Mon, 25 May 2009 09:58:54 +0700 Subject: [Freeswitch-users] Register SoftPhone behind NAT In-Reply-To: <4A19FDB0.5090200@telkom.co.id> References: <4aec92830905241409k48cd7430uc15516759c561989@mail.gmail.com> <4A19FDB0.5090200@telkom.co.id> Message-ID: <4A1A096E.7050309@telkom.co.id> Hi Guys, I implement FS as SIP Proxy including media (sip & media) , the server sitting behind NAT and the softphone client sitting behind NAT, so the configuration would be : voip provider (172.17.67.91) ---- (172.17.67.12) FS on VPN network (172.17.67.13) - NAT (222.192.12.23) public ----NAT (222.182.219.21) -- softphone client (behind nat). Anybody could share the profile configuration for softphone register behind NAT? thank you in advance, mashudi ***************************************** Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. From wasim at convergence.pk Sun May 24 20:20:36 2009 From: wasim at convergence.pk (Wasim Baig) Date: Mon, 25 May 2009 09:20:36 +0600 Subject: [Freeswitch-users] Register SoftPhone behind NAT In-Reply-To: <4A1A096E.7050309@telkom.co.id> References: <4aec92830905241409k48cd7430uc15516759c561989@mail.gmail.com> <4A19FDB0.5090200@telkom.co.id> <4A1A096E.7050309@telkom.co.id> Message-ID: On Mon, May 25, 2009 at 8:58 AM, mashudi wrote: Anybody could share the profile configuration for softphone register > behind NAT? http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA http://wiki.freeswitch.org/wiki/NAT http://wiki.freeswitch.org/wiki/NAT_Traversal http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/ac5c268a/attachment.html From jim at evolutiontel.net Sun May 24 20:23:39 2009 From: jim at evolutiontel.net (Jim Burke) Date: Mon, 25 May 2009 13:23:39 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> Message-ID: Hi Brian and Anthony, We need to move back a couple of steps here. I fully understand the A leg cannot enable SRTP unless it sends descriptors in the original INVITE. As the A party works as expected lets not discuss that any further as it clouds the waters so to speak. What I am trying to achieve is to set SRTP on a per leg basis if the UA requires it. In the case of terminating the B leg, if the UA requires SRTP, Freeswitch will not know this until advised by the B leg UA via a 415 Bad Security Level responce from the B leg INVITE. Per debug attached to original email, FS appears to generate the SRTP descriptors however does not add them to the second INVITE sent to the B leg. IMHO this is a fault and should be corrected. Anthony, do you have any thoughts on this! Call testing shows the following results. 1. A leg INVITE with SRTP descriptors in SDP and sip_secure_media set in the dialplan. B leg INVITE has no SRTP descriptors in SDP . RTP between A UA and FS uses SRTP, B leg does not. 2. A leg INVITE with SRTP descriptors in SDP and sip_secure_media and export sip_secure_media=true set in the dialplan. B leg INVITE also SRTP descriptors in SDP . RTP between A UA and FS uses SRTP, FS and B UA also uses SRTP. 3. A leg INVITE with no SRTP descriptors in SDP and export sip_secure_media=true set in the dialplan. B leg INVITE has SRTP descriptors in SDP. RTP between B UA and FS uses SRTP, A leg does not. 4. A leg INVITE without SRTP descriptors in SDP, B leg INVITE without SRTP descriptors in SDP results in 415 Bad Security Level. Dialplan set to continue on fail and export sip_secure_media=true then bridge the call once more. Debug shows that FS generates the SRTP descriptors, however FS does not add them to the second INVITE. As you can see from above, FS can set SRTP on a per leg basis. However for some reason it fails to add the SRTP descriptors to the SDP in the second INVITE for scenario 3. I hope this has cleared up the confusion regarding my original email. If you wish to discuss further please let me know what time the conference is and I can join in. Regards, Jim On Fri, May 22, 2009 at 11:59 PM, Brian West wrote: > > On May 22, 2009, at 12:47 AM, Jim Burke wrote: > > Hey Brian, > > Will have a look at ZRTP :) > > Not sure I understand your comments regarding its all over once > receiving the 415 from the B party. ?Is'nt that what parm > continue_on_fail does? ?The fact that it sends the invite back out > sorta proves this. > > The A-LEG has to hangup to re-enable SRTP it can't do it if it didn't invite > with it in the first place. > > The other point of interest here is that if you set application="export" data="sip_secure_media=true"/> before the first > bridge function it will include the security descriptions in the B leg > INVITE even when the A leg does not have them and the call will > succeed. ?The B Eyebeam will show the locked padlock while A does not. > > Make sure you do not answer the call before you do it. > > From what I can see in code it is this guy that must stop it all from > > happening. ?TFLAG_SECURE ?But I dont understand why :( > > Again you have to invite to FS with crypto it can't magically cause crypto > to work unless you initiate it with your first invite. > > Regards, > Jim > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jim at evolutiontel.net Sun May 24 20:27:13 2009 From: jim at evolutiontel.net (Jim Burke) Date: Mon, 25 May 2009 13:27:13 +1000 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> Message-ID: Congrats......So what did you decide on, this is your chance for a free plug ;) On Sun, May 24, 2009 at 6:31 AM, Diego Viola wrote: > Thanks everyone, we just choose our own name :). > > On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote: >> >> I just went through naming my own thing -- pretty happy with how it came >> out. ?I'm setting up a wisp for my 2 square mile city in los angeles metro >> area. >> The name: ?UNWIRE.IT >> http://unwire.it >> :-) >> The other one I was/am toying with is WEHO.FI >> (short for west hollywood) >> On Fri, May 22, 2009 at 8:51 AM, EdPimentl wrote: >>> >>> VoiceCLOUD >>> CLOUDvoice >>> GlobalVoice >>> VoiceUP >>> voicEVERYthing >>> VoicEnterprise >>> S/IP (Services over IP) >>> GlobalSIP >>> VoiPLATFORM >>> >>> >>> Best regards, >>> -E >>> Gpro.ws >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( >> broadband internet for west hollywood >> ? ?d at unwire.it ? ? ? ?http://unwire.it >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jim at evolutiontel.net Sun May 24 20:29:10 2009 From: jim at evolutiontel.net (Jim Burke) Date: Mon, 25 May 2009 13:29:10 +1000 Subject: [Freeswitch-users] Cool names for my VoIP company In-Reply-To: References: <86a32abc0905212126t2ea88c05w272a35de975f0835@mail.gmail.com> <7e2ac3270905220845r41fafb93g7aa46f708125b61f@mail.gmail.com> <9dc4a1670905220851j582365ccn6a25defe7c1454a5@mail.gmail.com> <989132e70905221057h49f0a613pae9215eb9afa662d@mail.gmail.com> <86a32abc0905231331i3a71c1f2gc83a215266ffc96e@mail.gmail.com> Message-ID: And the tag line could include something about big savings. On Sun, May 24, 2009 at 7:30 AM, Brian West wrote: > I vote for Moose Penis Telecom > > /b > > On May 23, 2009, at 3:31 PM, Diego Viola wrote: > >> Thanks everyone, we just choose our own name :). >> >> On Fri, May 22, 2009 at 1:57 PM, Darin Weeks wrote: >>> >>> I just went through naming my own thing -- pretty happy with how it >>> came >>> out. ?I'm setting up a wisp for my 2 square mile city in los >>> angeles metro >>> area. >>> The name: ?UNWIRE.IT >>> http://unwire.it >>> :-) >>> The other one I was/am toying with is WEHO.FI >>> (short for west hollywood) >>> On Fri, May 22, 2009 at 8:51 AM, EdPimentl >>> wrote: >>>> >>>> VoiceCLOUD >>>> CLOUDvoice >>>> GlobalVoice >>>> VoiceUP >>>> voicEVERYthing >>>> VoicEnterprise >>>> S/IP (Services over IP) >>>> GlobalSIP >>>> VoiPLATFORM >>>> >>>> >>>> Best regards, >>>> -E >>>> Gpro.ws >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( >>> broadband internet for west hollywood >>> ? ?d at unwire.it ? ? ? ?http://unwire.it >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mashudiflexi at telkom.co.id Sun May 24 21:25:48 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Mon, 25 May 2009 11:25:48 +0700 Subject: [Freeswitch-users] err after installling freeswitch In-Reply-To: <20090525021737.GA31033@jdc.jasonjgw.net> References: <557185.81756.qm@web34302.mail.mud.yahoo.com> <20090525021737.GA31033@jdc.jasonjgw.net> Message-ID: <4A1A1DCC.1050201@telkom.co.id> Hi Guys, I have install Freeswitch with version : FreeSWITCH Version 1.0.4pre7 (13238M) Started. I load the openzap module after install the wanpipe modul and everything running , after I query the status of the Sangoma Card (A104D quad E1) we got error like this : 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 switch_core_db_persistant_execute_trans() SQL ERR [unable to open database file] 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 switch_core_db_persistant_execute_trans() SQL ERR [unable to open database file] 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 switch_core_db_persistant_execute_trans() SQL ERR [unable to open database file] please help. mashudi ***************************************** Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. From fvillarroel at yahoo.com Sun May 24 21:48:00 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Sun, 24 May 2009 21:48:00 -0700 (PDT) Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED Message-ID: <820833.94846.qm@web34307.mail.mud.yahoo.com> Dear. Excuse some off topic of Wikipbx, but today i solved my problem; wikipbx and FS working now. Best regard. --- On Sun, 5/24/09, Jason White wrote: > From: Jason White > Subject: Re: [Freeswitch-users] FS & Wikipbx Help > To: freeswitch-users at lists.freeswitch.org > Date: Sunday, May 24, 2009, 11:17 PM > FERNANDO VILLARROEL > wrote: > > > My install of wikipbx was succesfully, but i have > problem for registering a > > Softphone Xlite for testing; i look the next warning > in FS_CLI: > > > > freeswitch at internal> 2009-04-28 06:13:58 [WARNING] > sofia_reg.c:1701 > > sofia_reg_parse_auth() Can't find user > [223 at 192.168.1.150] You must define a > > domain called '192.168.1.150' in your directory and > add a user with the > > id="223" attribute and you must configure your device > to use the proper > > domain in it's authentication credentials. > > > > When i solve this problem i want make a gateway > between FS and Asterisk, for > > sending traffic from FS to Asterisk. > > > > If any could help me to resolve register ATA or > Softphones with wikipbx. > > If WikiPBX has its own mailing list, I think you would > receive better help > there. > > After installing FreeSWITCH, try registering to one of the > extensions (1000 to > 1019) provided in the default configuration. If that works, > which it should, > you should then try creating your own extension, or > modifying an existing one > as required. > > The error message you quote says very explicitly what is > wrong, but without > access to your FreeSWITCH configuration, it is impossible > to help any further, > except to suggest that you start with the default > configuration and work from > there. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mashudiflexi at telkom.co.id Mon May 25 00:24:44 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Mon, 25 May 2009 14:24:44 +0700 Subject: [Freeswitch-users] how to disbale : switch_core_sqldb() Message-ID: <4A1A47BC.1020905@telkom.co.id> Hi Guys, How to disable process starting of sql DB when we starting FreeSwitch ? here is the log from starting FreeSwitch : 32m2009-05-25 14:00:18 [INFO] switch_core_sqldb.c:494 switch_core_sqldb_start() Opening DB thank you in advance, mashudi ***************************************** Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. From shaheryarkh at googlemail.com Mon May 25 02:42:27 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 25 May 2009 15:42:27 +0600 Subject: [Freeswitch-users] how to disbale : switch_core_sqldb() In-Reply-To: <4A1A47BC.1020905@telkom.co.id> References: <4A1A47BC.1020905@telkom.co.id> Message-ID: I think passing -nosql as argument to freeswitch start up command will do this. For example, bash#> freeswitch -hp -nosql -nc Thank you. On Mon, May 25, 2009 at 1:24 PM, mashudi wrote: > Hi Guys, > How to disable process starting of sql DB when we starting FreeSwitch ? > here is the log from starting FreeSwitch : > 32m2009-05-25 14:00:18 [INFO] switch_core_sqldb.c:494 > switch_core_sqldb_start() Opening DB > > thank you in advance, > > > mashudi > > ***************************************** > Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, > DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? > Ikuti Dahsyatnya FLEXI KOMUNITAS. > Ketik CREATE[NAMA GRUP], sms ke 345. > Contoh: CREATE SMU2, sms ke 345. > Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/e4154987/attachment-0001.html From simon at imaginator.com Mon May 25 03:13:18 2009 From: simon at imaginator.com (Simon Tennant) Date: Mon, 25 May 2009 12:13:18 +0200 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: References: <4A183F13.2030108@imaginator.com> <30153D4481AF4017B2A9DDE90320905B@voztovoice> <4A19002D.1080800@imaginator.com> <3CE3091789184A0AB909C026F9E8D8CA@voztovoice> Message-ID: <4A1A6F3E.9090109@imaginator.com> ...actually it's not "quite simple". Nokia handsets have a myriad of other options that help fine tune things like STUN server resolution and are only editable after installing some nokia supplied software. We've gone through all those steps and still calls will fail for odd reasons or more than one handset cannot access another handset behind the same NAT. Or handsets will have STUN resolution problems or caller-id problems. /me wishes SNOM made mobiles. S. Wasim Baig wrote: > On Sun, May 24, 2009 at 8:17 PM, bakko > wrote: > > I just installing FS in my new server. When i can make tests y > will write > you. > > > bakko: > > its really quite simple ... and should work straight out of the box > > first install your server - > http://wiki.freeswitch.org/wiki/Installation_Guide > then get started using the - > http://wiki.freeswitch.org/wiki/Getting_Started_Guide > then configure Nokia - http://wiki.freeswitch.org/wiki/Nokia_N95 > > -- > wasim h. baig | principal consultant | convergence pk | +92 300 > 8508070 | peace be upon you ... > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Simon Tennant _____________________________________________ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: simon at buddycloud.com From wasim at convergence.pk Mon May 25 03:25:41 2009 From: wasim at convergence.pk (Wasim Baig) Date: Mon, 25 May 2009 16:25:41 +0600 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: <4A1A6F3E.9090109@imaginator.com> References: <4A183F13.2030108@imaginator.com> <30153D4481AF4017B2A9DDE90320905B@voztovoice> <4A19002D.1080800@imaginator.com> <3CE3091789184A0AB909C026F9E8D8CA@voztovoice> <4A1A6F3E.9090109@imaginator.com> Message-ID: On Mon, May 25, 2009 at 4:13 PM, Simon Tennant wrote: > ...actually it's not "quite simple". Nokia handsets have a myriad of > other options that help fine tune things like STUN server resolution and > are only editable after installing some nokia supplied software. concur, but those are related to Nokia, and would affect any softswitch beyond the nat ... not just fs We've gone through all those steps and still calls will fail for odd > reasons or more than one handset cannot access another handset behind > the same NAT. Or handsets will have STUN resolution problems or > caller-id problems. my original point of it being simple was under the caveat of a simple setup, same lan, default configs, works out of the box ... > /me wishes SNOM made mobiles. indeed ... -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/af0d2ed4/attachment.html From saigop at gmail.com Mon May 25 04:27:17 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Mon, 25 May 2009 16:57:17 +0530 Subject: [Freeswitch-users] uuid_transfer gets break Message-ID: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> Hi, I had some discussion with the IRC regarding the uuid_transfer gets hang-up where the call is originated via javascript thru event socket. I was suggested to install latest SVN trunk. I did that again the same issue, the log is attached with here http://pastebin.freeswitch.org/9103 My call flow like this, 1. api jsrun fils.js 2. capture the uuid 3. api uuid_transfer -both Both the leg gets hangedup. Someone can assist me where I am wrong. -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/f6bd1f48/attachment.html From asannucci at gmail.com Mon May 25 05:59:15 2009 From: asannucci at gmail.com (bakko) Date: Mon, 25 May 2009 14:59:15 +0200 Subject: [Freeswitch-users] Nokia SIP clients: share your working config? In-Reply-To: References: <4A183F13.2030108@imaginator.com><30153D4481AF4017B2A9DDE90320905B@voztovoice> <4A19002D.1080800@imaginator.com><3CE3091789184A0AB909C026F9E8D8CA@voztovoice> Message-ID: <0786C531772446898854E845FEB5A436@voztovoice> Thank you, but with nokia 6300i i have to configure manualy the xml configuration file and then upload it tu my cellphone :) Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/6e4d5a56/attachment.html From odermann at googlemail.com Mon May 25 08:05:44 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 25 May 2009 17:05:44 +0200 Subject: [Freeswitch-users] Fax through FS to Callweaver. How? In-Reply-To: <4A0AE600.3040401@mctelefonia.com> References: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> <4A0AE600.3040401@mctelefonia.com> Message-ID: <5e414ed0905250805s27b2fc6eke53b282cbd2a07d3@mail.gmail.com> hi, sorry for the late reply and thanks for the replies. it working quite fine now. we still habe some problems, because we have 2 nic's (internal ip/external ip). we have to find a way through different ip-adresses, ip-areas, firewall and switch :-) or is there a way to make fs listen to two ip-adresses? kind regards dennis 2009/5/13 Antonio Gallo : > Dennis ha scritto: >> does someone know callweaver and can tell me, if there are some >> important settings to be set for making it work with fs in the middle? >> > Look at this, i needed to apply it using a Patton gateway too: > ? ?http://www.callweaver.org/ticket/487 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From odermann at googlemail.com Mon May 25 08:19:44 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 25 May 2009 17:19:44 +0200 Subject: [Freeswitch-users] How to work with the debug logfile? Message-ID: <5e414ed0905250819o792c42f9j66fe3c70730a89d0@mail.gmail.com> hi, we encounter some small problems withing the past 2 days and we are trying to find out more about the problems. for this we downloaded the debug logfiles written by fs, but we do not manage to filter all log-entries for one single special call. we are using socket outbound and would like to see all entries (inbound/outbound) of one call. is this possible? kind regards dennis From fvillarroel at yahoo.com Mon May 25 12:44:15 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 25 May 2009 12:44:15 -0700 (PDT) Subject: [Freeswitch-users] Calls drop at 30 seconds Message-ID: <917814.75765.qm@web34301.mail.mud.yahoo.com> Hi, I have 2 softphones (101 and 102) logged to my FS in a LAN, but the calls drop at 30 seconds: 2009-04-29 22:44:12 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/101 at 192.168.1.150 entering state [terminating][0] 2009-04-29 22:44:12 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/admin/101 at 192.168.1.150 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/101 at 192.168.1.150 [KILL] 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/101 at 192.168.1.150 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/101 at 192.168.1.150] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 audio_bridge_thread() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/admin/102 receive message [UNBRIDGE] 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/102] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 audio_bridge_thread() Send signal sofia/admin/101 at 192.168.1.150 [BREAK] 2009-04-29 22:44:12 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup sofia/admin/102 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/102 [KILL] 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:493 switch_core_session_run() (sofia/admin/102) State EXCHANGE_MEDIA going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/102) Running State Change CS_HANGUP 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/102) State HANGUP 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/102 hanging up, cause: NORMAL_CLEARING 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending BYE to sofia/admin/102 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/102 Standard HANGUP, cause: NORMAL_CLEARING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/102) State HANGUP going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/102) State Change CS_HANGUP -> CS_REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/102) Running State Change CS_REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/102) State REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State EXECUTE going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) Running State Change CS_HANGUP 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State HANGUP 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/101 at 192.168.1.150 hanging up, cause: NORMAL_UNSPECIFIED 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/101 at 192.168.1.150 Standard HANGUP, cause: NORMAL_UNSPECIFIED 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State HANGUP going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State Change CS_HANGUP -> CS_REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/101 at 192.168.1.150 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) Running State Change CS_REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/101 at 192.168.1.150) State REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/101 at 192.168.1.150 Standard REPORTING, cause: NORMAL_UNSPECIFIED 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/101 at 192.168.1.150) State REPORTING going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State Change CS_REPORTING -> CS_DESTROY 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 35 (sofia/admin/101 at 192.168.1.150) Locked, Waiting on external entities 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 35 (sofia/admin/101 at 192.168.1.150) Ended 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/101 at 192.168.1.150 [CS_DESTROY] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/101 at 192.168.1.150) State DESTROY 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/101 at 192.168.1.150 SOFIA DESTROY 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/101 at 192.168.1.150 Standard DESTROY 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/101 at 192.168.1.150) State DESTROY going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/102 Standard REPORTING, cause: NORMAL_CLEARING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/102) State REPORTING going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/102) State Change CS_REPORTING -> CS_DESTROY 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 36 (sofia/admin/102) Locked, Waiting on external entities 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 36 (sofia/admin/102) Ended 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/102 [CS_DESTROY] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/102) State DESTROY 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/102 SOFIA DESTROY 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/102 Standard DESTROY 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/102) State DESTROY going to sleep Any idea. Fernando Villarroel. From diego.viola at gmail.com Mon May 25 13:13:11 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 25 May 2009 20:13:11 +0000 Subject: [Freeswitch-users] Calls drop at 30 seconds In-Reply-To: <917814.75765.qm@web34301.mail.mud.yahoo.com> References: <917814.75765.qm@web34301.mail.mud.yahoo.com> Message-ID: <86a32abc0905251313l212479c2lbdfc35e58a3a724@mail.gmail.com> I had the same issue before, and it was a LAN problem, make sure your network is configured properly. Are you running the softphones and FS on the same machine? Diego On Mon, May 25, 2009 at 7:44 PM, FERNANDO VILLARROEL wrote: > > Hi, > > I have 2 softphones (101 and 102) logged to my FS in a LAN, but the calls drop at 30 seconds: > > 2009-04-29 22:44:12 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/101 at 192.168.1.150 entering state [terminating][0] > 2009-04-29 22:44:12 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/admin/101 at 192.168.1.150 [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/101 at 192.168.1.150 [KILL] > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/101 at 192.168.1.150 [BREAK] > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/101 at 192.168.1.150] > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 audio_bridge_thread() Send signal sofia/admin/102 [BREAK] > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/admin/102 receive message [UNBRIDGE] > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/admin/102 [BREAK] > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/102] > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 audio_bridge_thread() Send signal sofia/admin/101 at 192.168.1.150 [BREAK] > 2009-04-29 22:44:12 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup sofia/admin/102 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/102 [KILL] > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/102 [BREAK] > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:493 switch_core_session_run() (sofia/admin/102) State EXCHANGE_MEDIA going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/102) Running State Change CS_HANGUP > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/102) State HANGUP > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/102 hanging up, cause: NORMAL_CLEARING > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending BYE to sofia/admin/102 > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/102 Standard HANGUP, cause: NORMAL_CLEARING > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/102) State HANGUP going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/102) State Change CS_HANGUP -> CS_REPORTING > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/102 [BREAK] > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/102) Running State Change CS_REPORTING > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/102) State REPORTING > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State EXECUTE going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) Running State Change CS_HANGUP > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State HANGUP > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/101 at 192.168.1.150 hanging up, cause: NORMAL_UNSPECIFIED > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/101 at 192.168.1.150 Standard HANGUP, cause: NORMAL_UNSPECIFIED > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State HANGUP going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State Change CS_HANGUP -> CS_REPORTING > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/101 at 192.168.1.150 [BREAK] > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) Running State Change CS_REPORTING > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/101 at 192.168.1.150) State REPORTING > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/101 at 192.168.1.150 Standard REPORTING, cause: NORMAL_UNSPECIFIED > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/101 at 192.168.1.150) State REPORTING going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/101 at 192.168.1.150) State Change CS_REPORTING -> CS_DESTROY > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 35 (sofia/admin/101 at 192.168.1.150) Locked, Waiting on external entities > 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 35 (sofia/admin/101 at 192.168.1.150) Ended > 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/101 at 192.168.1.150 [CS_DESTROY] > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/101 at 192.168.1.150) State DESTROY > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/101 at 192.168.1.150 SOFIA DESTROY > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/101 at 192.168.1.150 Standard DESTROY > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/101 at 192.168.1.150) State DESTROY going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/admin/102 Standard REPORTING, cause: NORMAL_CLEARING > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/102) State REPORTING going to sleep > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/admin/102) State Change CS_REPORTING -> CS_DESTROY > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 36 (sofia/admin/102) Locked, Waiting on external entities > 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 36 (sofia/admin/102) Ended > 2009-04-29 22:44:12 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/admin/102 [CS_DESTROY] > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/102) State DESTROY > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/admin/102 SOFIA DESTROY > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/admin/102 Standard DESTROY > 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/admin/102) State DESTROY going to sleep > > > Any idea. > > Fernando Villarroel. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fvillarroel at yahoo.com Mon May 25 13:31:47 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 25 May 2009 13:31:47 -0700 (PDT) Subject: [Freeswitch-users] Calls drop at 30 seconds Message-ID: <879169.27282.qm@web34306.mail.mud.yahoo.com> Hi Diego, The softphones are in different machines (Softphone 1 Win XP and Softphone 2 in other win XP), i have ringback, but no audio and call death at 30 seconds. Fernando --- On Mon, 5/25/09, Diego Viola wrote: > From: Diego Viola > Subject: Re: [Freeswitch-users] Calls drop at 30 seconds > To: freeswitch-users at lists.freeswitch.org > Date: Monday, May 25, 2009, 5:13 PM > I had the same issue before, and it > was a LAN problem, make sure your > network is configured properly. > > Are you running the softphones and FS on the same machine? > > Diego > > On Mon, May 25, 2009 at 7:44 PM, FERNANDO VILLARROEL > > wrote: > > > > Hi, > > > > I have 2 softphones (101 and 102) logged to my FS in a > LAN, but the calls drop at 30 seconds: > > > > 2009-04-29 22:44:12 [DEBUG] sofia.c:3037 > sofia_handle_sip_i_state() Channel > sofia/admin/101 at 192.168.1.150 entering state > [terminating][0] > > 2009-04-29 22:44:12 [NOTICE] sofia.c:3597 > sofia_handle_sip_i_state() Hangup > sofia/admin/101 at 192.168.1.150 [CS_EXECUTE] > [NORMAL_UNSPECIFIED] > > 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/admin/101 at 192.168.1.150 [KILL] > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/101 at 192.168.1.150 [BREAK] > > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 > audio_bridge_thread() BRIDGE THREAD DONE > [sofia/admin/101 at 192.168.1.150] > > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 > audio_bridge_thread() Send signal sofia/admin/102 [BREAK] > > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:426 > audio_bridge_thread() sofia/admin/102 receive message > [UNBRIDGE] > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:630 > switch_core_session_perform_receive_message() Send signal > sofia/admin/102 [BREAK] > > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 > audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/102] > > 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 > audio_bridge_thread() Send signal > sofia/admin/101 at 192.168.1.150 [BREAK] > > 2009-04-29 22:44:12 [NOTICE] switch_ivr_bridge.c:505 > audio_bridge_on_exchange_media() Hangup sofia/admin/102 > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal sofia/admin/102 > [KILL] > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/102 [BREAK] > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:493 switch_core_session_run() > (sofia/admin/102) State EXCHANGE_MEDIA going to sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:397 switch_core_session_run() > (sofia/admin/102) Running State Change CS_HANGUP > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:433 switch_core_session_run() > (sofia/admin/102) State HANGUP > > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 > sofia_on_hangup() Channel sofia/admin/102 hanging up, cause: > NORMAL_CLEARING > > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:378 > sofia_on_hangup() Sending BYE to sofia/admin/102 > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/admin/102 Standard > HANGUP, cause: NORMAL_CLEARING > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:433 switch_core_session_run() > (sofia/admin/102) State HANGUP going to sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:475 switch_core_session_run() > (sofia/admin/102) State Change CS_HANGUP -> CS_REPORTING > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/102 [BREAK] > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:397 switch_core_session_run() > (sofia/admin/102) Running State Change CS_REPORTING > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/102) > State REPORTING > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:490 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) State EXECUTE going to > sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:397 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) Running State Change > CS_HANGUP > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:433 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) State HANGUP > > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 > sofia_on_hangup() Channel sofia/admin/101 at 192.168.1.150 > hanging up, cause: NORMAL_UNSPECIFIED > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:46 > switch_core_standard_on_hangup() > sofia/admin/101 at 192.168.1.150 Standard HANGUP, cause: > NORMAL_UNSPECIFIED > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:433 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) State HANGUP going to sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:475 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) State Change CS_HANGUP -> > CS_REPORTING > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/admin/101 at 192.168.1.150 [BREAK] > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:397 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) Running State Change > CS_REPORTING > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/101 at 192.168.1.150) State REPORTING > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:53 > switch_core_standard_on_reporting() > sofia/admin/101 at 192.168.1.150 Standard REPORTING, cause: > NORMAL_UNSPECIFIED > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:607 > switch_core_session_reporting_state() > (sofia/admin/101 at 192.168.1.150) State REPORTING going to > sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:410 switch_core_session_run() > (sofia/admin/101 at 192.168.1.150) State Change CS_REPORTING > -> CS_DESTROY > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 35 > (sofia/admin/101 at 192.168.1.150) Locked, Waiting on external > entities > > 2009-04-29 22:44:12 [NOTICE] > switch_core_session.c:1085 switch_core_session_thread() > Session 35 (sofia/admin/101 at 192.168.1.150) Ended > > 2009-04-29 22:44:12 [NOTICE] > switch_core_session.c:1087 switch_core_session_thread() > Close Channel sofia/admin/101 at 192.168.1.150 [CS_DESTROY] > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/101 at 192.168.1.150) State DESTROY > > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 > sofia_on_destroy() sofia/admin/101 at 192.168.1.150 SOFIA > DESTROY > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:60 > switch_core_standard_on_destroy() > sofia/admin/101 at 192.168.1.150 Standard DESTROY > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:559 > switch_core_session_destroy_state() > (sofia/admin/101 at 192.168.1.150) State DESTROY going to > sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/admin/102 Standard > REPORTING, cause: NORMAL_CLEARING > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/admin/102) > State REPORTING going to sleep > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:410 switch_core_session_run() > (sofia/admin/102) State Change CS_REPORTING -> > CS_DESTROY > > 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 36 (sofia/admin/102) > Locked, Waiting on external entities > > 2009-04-29 22:44:12 [NOTICE] > switch_core_session.c:1085 switch_core_session_thread() > Session 36 (sofia/admin/102) Ended > > 2009-04-29 22:44:12 [NOTICE] > switch_core_session.c:1087 switch_core_session_thread() > Close Channel sofia/admin/102 [CS_DESTROY] > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/102) State > DESTROY > > 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 > sofia_on_destroy() sofia/admin/102 SOFIA DESTROY > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/admin/102 Standard > DESTROY > > 2009-04-29 22:44:12 [DEBUG] > switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/admin/102) State > DESTROY going to sleep > > > > > > Any idea. > > > > Fernando Villarroel. > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Mon May 25 14:08:30 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 25 May 2009 23:08:30 +0200 Subject: [Freeswitch-users] Calls drop at 30 seconds In-Reply-To: <879169.27282.qm@web34306.mail.mud.yahoo.com> References: <879169.27282.qm@web34306.mail.mud.yahoo.com> Message-ID: <4A1B08CE.4080008@gmx.net> I had a similar behaviour with dropped calls. After I changed the firewall on the FS machine it worked. In my case some ports on the FS machine were not open for outbound traffic (inbound were ok). Check SIP, TLS, RTP, STUN, DNS ports. Best regards Peter FERNANDO VILLARROEL schrieb: > Hi Diego, > > The softphones are in different machines (Softphone 1 Win XP and Softphone 2 in other win XP), i have ringback, but no audio and call death at 30 seconds. > > Fernando > > --- On Mon, 5/25/09, Diego Viola wrote: > > >> From: Diego Viola >> Subject: Re: [Freeswitch-users] Calls drop at 30 seconds >> To: freeswitch-users at lists.freeswitch.org >> Date: Monday, May 25, 2009, 5:13 PM >> I had the same issue before, and it >> was a LAN problem, make sure your >> network is configured properly. >> >> Are you running the softphones and FS on the same machine? >> >> Diego >> >> On Mon, May 25, 2009 at 7:44 PM, FERNANDO VILLARROEL >> >> wrote: >> >>> Hi, >>> >>> I have 2 softphones (101 and 102) logged to my FS in a >>> >> LAN, but the calls drop at 30 seconds: >> >>> 2009-04-29 22:44:12 [DEBUG] sofia.c:3037 >>> >> sofia_handle_sip_i_state() Channel >> sofia/admin/101 at 192.168.1.150 entering state >> [terminating][0] >> >>> 2009-04-29 22:44:12 [NOTICE] sofia.c:3597 >>> >> sofia_handle_sip_i_state() Hangup >> sofia/admin/101 at 192.168.1.150 [CS_EXECUTE] >> [NORMAL_UNSPECIFIED] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 >>> >> switch_channel_perform_hangup() Send signal >> sofia/admin/101 at 192.168.1.150 [KILL] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >>> >> switch_core_session_signal_state_change() Send signal >> sofia/admin/101 at 192.168.1.150 [BREAK] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 >>> >> audio_bridge_thread() BRIDGE THREAD DONE >> [sofia/admin/101 at 192.168.1.150] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 >>> >> audio_bridge_thread() Send signal sofia/admin/102 [BREAK] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:426 >>> >> audio_bridge_thread() sofia/admin/102 receive message >> [UNBRIDGE] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:630 >>> >> switch_core_session_perform_receive_message() Send signal >> sofia/admin/102 [BREAK] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 >>> >> audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/102] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 >>> >> audio_bridge_thread() Send signal >> sofia/admin/101 at 192.168.1.150 [BREAK] >> >>> 2009-04-29 22:44:12 [NOTICE] switch_ivr_bridge.c:505 >>> >> audio_bridge_on_exchange_media() Hangup sofia/admin/102 >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 >>> >> switch_channel_perform_hangup() Send signal sofia/admin/102 >> [KILL] >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >>> >> switch_core_session_signal_state_change() Send signal >> sofia/admin/102 [BREAK] >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:493 switch_core_session_run() >> (sofia/admin/102) State EXCHANGE_MEDIA going to sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/102) Running State Change CS_HANGUP >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/102) State HANGUP >> >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 >>> >> sofia_on_hangup() Channel sofia/admin/102 hanging up, cause: >> NORMAL_CLEARING >> >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:378 >>> >> sofia_on_hangup() Sending BYE to sofia/admin/102 >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() sofia/admin/102 Standard >> HANGUP, cause: NORMAL_CLEARING >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/102) State HANGUP going to sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:475 switch_core_session_run() >> (sofia/admin/102) State Change CS_HANGUP -> CS_REPORTING >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >>> >> switch_core_session_signal_state_change() Send signal >> sofia/admin/102 [BREAK] >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/102) Running State Change CS_REPORTING >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/admin/102) >> State REPORTING >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:490 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State EXECUTE going to >> sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) Running State Change >> CS_HANGUP >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State HANGUP >> >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 >>> >> sofia_on_hangup() Channel sofia/admin/101 at 192.168.1.150 >> hanging up, cause: NORMAL_UNSPECIFIED >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() >> sofia/admin/101 at 192.168.1.150 Standard HANGUP, cause: >> NORMAL_UNSPECIFIED >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State HANGUP going to sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:475 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State Change CS_HANGUP -> >> CS_REPORTING >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >>> >> switch_core_session_signal_state_change() Send signal >> sofia/admin/101 at 192.168.1.150 [BREAK] >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) Running State Change >> CS_REPORTING >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() >> (sofia/admin/101 at 192.168.1.150) State REPORTING >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() >> sofia/admin/101 at 192.168.1.150 Standard REPORTING, cause: >> NORMAL_UNSPECIFIED >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() >> (sofia/admin/101 at 192.168.1.150) State REPORTING going to >> sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:410 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State Change CS_REPORTING >> -> CS_DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 >>> >> switch_core_session_thread() Session 35 >> (sofia/admin/101 at 192.168.1.150) Locked, Waiting on external >> entities >> >>> 2009-04-29 22:44:12 [NOTICE] >>> >> switch_core_session.c:1085 switch_core_session_thread() >> Session 35 (sofia/admin/101 at 192.168.1.150) Ended >> >>> 2009-04-29 22:44:12 [NOTICE] >>> >> switch_core_session.c:1087 switch_core_session_thread() >> Close Channel sofia/admin/101 at 192.168.1.150 [CS_DESTROY] >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() >> (sofia/admin/101 at 192.168.1.150) State DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 >>> >> sofia_on_destroy() sofia/admin/101 at 192.168.1.150 SOFIA >> DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() >> sofia/admin/101 at 192.168.1.150 Standard DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() >> (sofia/admin/101 at 192.168.1.150) State DESTROY going to >> sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() sofia/admin/102 Standard >> REPORTING, cause: NORMAL_CLEARING >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/admin/102) >> State REPORTING going to sleep >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:410 switch_core_session_run() >> (sofia/admin/102) State Change CS_REPORTING -> >> CS_DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 >>> >> switch_core_session_thread() Session 36 (sofia/admin/102) >> Locked, Waiting on external entities >> >>> 2009-04-29 22:44:12 [NOTICE] >>> >> switch_core_session.c:1085 switch_core_session_thread() >> Session 36 (sofia/admin/102) Ended >> >>> 2009-04-29 22:44:12 [NOTICE] >>> >> switch_core_session.c:1087 switch_core_session_thread() >> Close Channel sofia/admin/102 [CS_DESTROY] >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/admin/102) State >> DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 >>> >> sofia_on_destroy() sofia/admin/102 SOFIA DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() sofia/admin/102 Standard >> DESTROY >> >>> 2009-04-29 22:44:12 [DEBUG] >>> >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/admin/102) State >> DESTROY going to sleep >> >>> Any idea. >>> >>> Fernando Villarroel. >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From krice at freeswitch.org Mon May 25 14:24:21 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 May 2009 16:24:21 -0500 Subject: [Freeswitch-users] Calls drop at 30 seconds In-Reply-To: <879169.27282.qm@web34306.mail.mud.yahoo.com> Message-ID: Sounds like lack of RTP ports being open due to iptables or firewall... 30 seconds sounds like the RTP timeout being hit... > From: FERNANDO VILLARROEL > Reply-To: > Date: Mon, 25 May 2009 13:31:47 -0700 (PDT) > To: > Subject: Re: [Freeswitch-users] Calls drop at 30 seconds > > > Hi Diego, > > The softphones are in different machines (Softphone 1 Win XP and Softphone 2 > in other win XP), i have ringback, but no audio and call death at 30 seconds. > > Fernando > > --- On Mon, 5/25/09, Diego Viola wrote: > >> From: Diego Viola >> Subject: Re: [Freeswitch-users] Calls drop at 30 seconds >> To: freeswitch-users at lists.freeswitch.org >> Date: Monday, May 25, 2009, 5:13 PM >> I had the same issue before, and it >> was a LAN problem, make sure your >> network is configured properly. >> >> Are you running the softphones and FS on the same machine? >> >> Diego >> >> On Mon, May 25, 2009 at 7:44 PM, FERNANDO VILLARROEL >> >> wrote: >>> >>> Hi, >>> >>> I have 2 softphones (101 and 102) logged to my FS in a >> LAN, but the calls drop at 30 seconds: >>> >>> 2009-04-29 22:44:12 [DEBUG] sofia.c:3037 >> sofia_handle_sip_i_state() Channel >> sofia/admin/101 at 192.168.1.150 entering state >> [terminating][0] >>> 2009-04-29 22:44:12 [NOTICE] sofia.c:3597 >> sofia_handle_sip_i_state() Hangup >> sofia/admin/101 at 192.168.1.150 [CS_EXECUTE] >> [NORMAL_UNSPECIFIED] >>> 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 >> switch_channel_perform_hangup() Send signal >> sofia/admin/101 at 192.168.1.150 [KILL] >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/admin/101 at 192.168.1.150 [BREAK] >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 >> audio_bridge_thread() BRIDGE THREAD DONE >> [sofia/admin/101 at 192.168.1.150] >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 >> audio_bridge_thread() Send signal sofia/admin/102 [BREAK] >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:426 >> audio_bridge_thread() sofia/admin/102 receive message >> [UNBRIDGE] >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:630 >> switch_core_session_perform_receive_message() Send signal >> sofia/admin/102 [BREAK] >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 >> audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/102] >>> 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 >> audio_bridge_thread() Send signal >> sofia/admin/101 at 192.168.1.150 [BREAK] >>> 2009-04-29 22:44:12 [NOTICE] switch_ivr_bridge.c:505 >> audio_bridge_on_exchange_media() Hangup sofia/admin/102 >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 >> switch_channel_perform_hangup() Send signal sofia/admin/102 >> [KILL] >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/admin/102 [BREAK] >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:493 switch_core_session_run() >> (sofia/admin/102) State EXCHANGE_MEDIA going to sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/102) Running State Change CS_HANGUP >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/102) State HANGUP >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 >> sofia_on_hangup() Channel sofia/admin/102 hanging up, cause: >> NORMAL_CLEARING >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:378 >> sofia_on_hangup() Sending BYE to sofia/admin/102 >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() sofia/admin/102 Standard >> HANGUP, cause: NORMAL_CLEARING >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/102) State HANGUP going to sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:475 switch_core_session_run() >> (sofia/admin/102) State Change CS_HANGUP -> CS_REPORTING >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/admin/102 [BREAK] >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/102) Running State Change CS_REPORTING >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/admin/102) >> State REPORTING >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:490 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State EXECUTE going to >> sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) Running State Change >> CS_HANGUP >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State HANGUP >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 >> sofia_on_hangup() Channel sofia/admin/101 at 192.168.1.150 >> hanging up, cause: NORMAL_UNSPECIFIED >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() >> sofia/admin/101 at 192.168.1.150 Standard HANGUP, cause: >> NORMAL_UNSPECIFIED >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:433 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State HANGUP going to sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:475 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State Change CS_HANGUP -> >> CS_REPORTING >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/admin/101 at 192.168.1.150 [BREAK] >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:397 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) Running State Change >> CS_REPORTING >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() >> (sofia/admin/101 at 192.168.1.150) State REPORTING >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() >> sofia/admin/101 at 192.168.1.150 Standard REPORTING, cause: >> NORMAL_UNSPECIFIED >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() >> (sofia/admin/101 at 192.168.1.150) State REPORTING going to >> sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:410 switch_core_session_run() >> (sofia/admin/101 at 192.168.1.150) State Change CS_REPORTING >> -> CS_DESTROY >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 >> switch_core_session_thread() Session 35 >> (sofia/admin/101 at 192.168.1.150) Locked, Waiting on external >> entities >>> 2009-04-29 22:44:12 [NOTICE] >> switch_core_session.c:1085 switch_core_session_thread() >> Session 35 (sofia/admin/101 at 192.168.1.150) Ended >>> 2009-04-29 22:44:12 [NOTICE] >> switch_core_session.c:1087 switch_core_session_thread() >> Close Channel sofia/admin/101 at 192.168.1.150 [CS_DESTROY] >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() >> (sofia/admin/101 at 192.168.1.150) State DESTROY >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 >> sofia_on_destroy() sofia/admin/101 at 192.168.1.150 SOFIA >> DESTROY >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() >> sofia/admin/101 at 192.168.1.150 Standard DESTROY >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() >> (sofia/admin/101 at 192.168.1.150) State DESTROY going to >> sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() sofia/admin/102 Standard >> REPORTING, cause: NORMAL_CLEARING >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/admin/102) >> State REPORTING going to sleep >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:410 switch_core_session_run() >> (sofia/admin/102) State Change CS_REPORTING -> >> CS_DESTROY >>> 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:1067 >> switch_core_session_thread() Session 36 (sofia/admin/102) >> Locked, Waiting on external entities >>> 2009-04-29 22:44:12 [NOTICE] >> switch_core_session.c:1085 switch_core_session_thread() >> Session 36 (sofia/admin/102) Ended >>> 2009-04-29 22:44:12 [NOTICE] >> switch_core_session.c:1087 switch_core_session_thread() >> Close Channel sofia/admin/102 [CS_DESTROY] >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/admin/102) State >> DESTROY >>> 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:240 >> sofia_on_destroy() sofia/admin/102 SOFIA DESTROY >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() sofia/admin/102 Standard >> DESTROY >>> 2009-04-29 22:44:12 [DEBUG] >> switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/admin/102) State >> DESTROY going to sleep >>> >>> >>> Any idea. >>> >>> Fernando Villarroel. >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Mon May 25 14:29:09 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 25 May 2009 23:29:09 +0200 Subject: [Freeswitch-users] uuid_chat Message-ID: <4A1B0DA5.2090504@gmx.net> Hello, today I tried uuid_chat via event socket. A simple chat application works: "bgapi chat sip|agent1 at fqdn|agent1 at fqdn|Message". "uuid_chat " however returned "+OK", but nothing happens. Neither is there a debug line on the console, nor a SIP (in my case TLS) message is sent to the UA. Has anybody successfully tried this command and has some additional hints? Is there any further configuration needed? Best regards Peter From saeedahmad1981 at gmail.com Mon May 25 14:38:43 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Mon, 25 May 2009 23:38:43 +0200 Subject: [Freeswitch-users] xml cdr parser (PHP + MySQL) Message-ID: Hi all, Is there any script available which can be used to load xml cdr from log dir into mysql db? I found some scripts in /contrib but they are not loading it in db. I would really appreciate if someone can share their script. Many Thanks Saeed Ahmed. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/9a3437a3/attachment.html From evilla at chipoly.com Mon May 25 15:04:22 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Mon, 25 May 2009 16:04:22 -0600 Subject: [Freeswitch-users] FS in Amazon EC2 for production? Message-ID: <011301c9dd84$c3053610$490fa230$@com> Hello my friends. Has anyone used the EC2 for production? Tests? I'm wondering if it would be "better" to have a FS system in the cloud for carrier-to-carrier connections. Any ideas will be appreciated Thanks 2 u all Edwin Villarreal World Net Commerce SA CV WNC Telecom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/7cba384c/attachment.html From Prometheus001 at gmx.net Mon May 25 16:43:34 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 26 May 2009 01:43:34 +0200 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <011301c9dd84$c3053610$490fa230$@com> References: <011301c9dd84$c3053610$490fa230$@com> Message-ID: <4A1B2D26.90202@gmx.net> We have used FS on ec2 for testing purposes only. It was ok. We havn't done any performance test though. Best regards Peter Ing. Edwin Villarreal schrieb: > > Hello my friends. > > > > Has anyone used the EC2 for production? Tests? > > > > I?m wondering if it would be ?better? to have a FS system in the cloud > for carrier-to-carrier connections. > > > > Any ideas will be appreciated > > > > Thanks 2 u all > > > > *Edwin Villarreal* > > World Net Commerce SA CV > > WNC Telecom > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brad.tuan at gmail.com Sun May 24 22:04:45 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Mon, 25 May 2009 13:04:45 +0800 Subject: [Freeswitch-users] 407 Proxy Authentication Required In-Reply-To: References: Message-ID: I'm trying to pass a sip call from ProxyA to my Freeswitch but my FS always return a "407 Proxy Authentication Required" to ProxyA as follow Source Destination Protocol Info ProxyA MyFS SIP/SDP Request: INVITE sip:97730009 at xxx.xxx.xxx.xxx:5060;transport=udp, with session description MyFS ProxyA SIP Status: 100 Trying MyFS ProxyA SIP Status: 407 Proxy Authentication Required I have already tried to 1. sip_profiles/internal.xml,external.xml -> set auth- calls=false <<----It isn't work 2. comment <<----It isn't work,neither then add this line: and add a new list to the "acl.conf.xml" like So , Anyone please tell me how can i do?? Tkanks a lot -Brad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090525/90a41843/attachment.html From jens at vegeby.nu Mon May 25 14:13:43 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Mon, 25 May 2009 23:13:43 +0200 Subject: [Freeswitch-users] Fax through FS to Callweaver. How? In-Reply-To: <5e414ed0905250805s27b2fc6eke53b282cbd2a07d3@mail.gmail.com> References: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> <4A0AE600.3040401@mctelefonia.com> <5e414ed0905250805s27b2fc6eke53b282cbd2a07d3@mail.gmail.com> Message-ID: <30ee97110905251413k18c2a046q755938834d2fa3fd@mail.gmail.com> I think you can do that by creating another profile. Regards Jens On 5/25/09, Dennis wrote: > hi, > > sorry for the late reply and thanks for the replies. > > it working quite fine now. we still habe some problems, because we > have 2 nic's (internal ip/external ip). we have to find a way through > different ip-adresses, ip-areas, firewall and switch :-) > > or is there a way to make fs listen to two ip-adresses? > > > kind regards > dennis > > > > 2009/5/13 Antonio Gallo : >> Dennis ha scritto: >>> does someone know callweaver and can tell me, if there are some >>> important settings to be set for making it work with fs in the middle? >>> >> Look at this, i needed to apply it using a Patton gateway too: >> ? ?http://www.callweaver.org/ticket/487 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device Mvh/Regards Jens From mike at jerris.com Mon May 25 18:17:34 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 25 May 2009 21:17:34 -0400 Subject: [Freeswitch-users] err after installling freeswitch In-Reply-To: <4A1A1DCC.1050201@telkom.co.id> References: <557185.81756.qm@web34302.mail.mud.yahoo.com> <20090525021737.GA31033@jdc.jasonjgw.net> <4A1A1DCC.1050201@telkom.co.id> Message-ID: This looks like a permissions error on creating or opening files in the db directory. Mike On May 25, 2009, at 12:25 AM, mashudi wrote: > Hi Guys, > I have install Freeswitch with version : FreeSWITCH Version 1.0.4pre7 > (13238M) Started. > I load the openzap module after install the wanpipe modul and > everything > running , after I query the status of the Sangoma Card (A104D quad E1) > we got error like this : > 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 > switch_core_db_persistant_execute_trans() SQL ERR [unable to open > database file] > 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 > switch_core_db_persistant_execute_trans() SQL ERR [unable to open > database file] > 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 > switch_core_db_persistant_execute_trans() SQL ERR [unable to open > database file] > From jason at jasonjgw.net Mon May 25 18:31:28 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 26 May 2009 11:31:28 +1000 Subject: [Freeswitch-users] Fax through FS to Callweaver. How? In-Reply-To: <30ee97110905251413k18c2a046q755938834d2fa3fd@mail.gmail.com> References: <5e414ed0905130522v61451228ld3ac8a7d26effafa@mail.gmail.com> <4A0AE600.3040401@mctelefonia.com> <5e414ed0905250805s27b2fc6eke53b282cbd2a07d3@mail.gmail.com> <30ee97110905251413k18c2a046q755938834d2fa3fd@mail.gmail.com> Message-ID: <20090526013127.GA15335@jdc.jasonjgw.net> Jens Vegeby wrote: > I think you can do that by creating another profile. Then you can bind it to a specific IP address. From jason at jasonjgw.net Mon May 25 18:40:12 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 26 May 2009 11:40:12 +1000 Subject: [Freeswitch-users] 407 Proxy Authentication Required In-Reply-To: References: Message-ID: <20090526014012.GA16268@jdc.jasonjgw.net> Brad Tuan wrote: > So , Anyone please tell me how can i do?? Turning off the ACL and setting auth-calls to false should be enough to do it. To find out where the problem is in your configuration, set the log level to debug if it isn't already, and read the logs carefully. You can also turn on SIP tracing if necessary. From mashudiflexi at telkom.co.id Mon May 25 19:57:26 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Tue, 26 May 2009 09:57:26 +0700 Subject: [Freeswitch-users] how to disbale : switch_core_sqldb() In-Reply-To: References: <4A1A47BC.1020905@telkom.co.id> Message-ID: <4A1B5A96.8060003@telkom.co.id> Dear Muhammad Shahzad , it is work and the err after has gone, thank you very much for your suggestion. Muhammad Shahzad wrote: > I think passing -nosql as argument to freeswitch start up command will > do this. For example, > > bash#> freeswitch -hp -nosql -nc > > Thank you. > > > On Mon, May 25, 2009 at 1:24 PM, mashudi > wrote: > > Hi Guys, > How to disable process starting of sql DB when we starting > FreeSwitch ? > here is the log from starting FreeSwitch : > 32m2009-05-25 14:00:18 [INFO] switch_core_sqldb.c:494 > switch_core_sqldb_start() Opening DB > > thank you in advance, > > > mashudi > > ***************************************** > Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, > DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? > Ikuti Dahsyatnya FLEXI KOMUNITAS. > Ketik CREATE[NAMA GRUP], sms ke 345. > Contoh: CREATE SMU2, sms ke 345. > Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > ***************************************** > Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, > DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? > Ikuti Dahsyatnya FLEXI KOMUNITAS. > Ketik CREATE[NAMA GRUP], sms ke 345. > Contoh: CREATE SMU2, sms ke 345. > Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ***************************************** Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. From jim at evolutiontel.net Mon May 25 22:52:17 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 26 May 2009 15:52:17 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> Message-ID: Hi Gents, After much testing today, I have found the answer to this question. The following dialplan works as expected and the B leg is encryted while the A leg is not. As I wanted to bypass_media unless SRTP was enforced by either party I found that FS would not set bypass_media to false unless pre_answer was added. FYI...According to Counterpath ZRTP is not added to the retail versions of Eyebeam or X-Lite, you need to purchase a bulk order. Regards, Jim On Mon, May 25, 2009 at 1:23 PM, Jim Burke wrote: > Hi Brian and Anthony, > > We need to move back a couple of steps here. ?I fully understand the A > leg cannot enable SRTP unless it sends descriptors in the original > INVITE. ?As the A party works as expected lets not discuss that any > further as it clouds the waters so to speak. > > What I am trying to achieve is to set SRTP on a per leg basis if the > UA requires it. ?In the case of terminating the B leg, if the UA > requires SRTP, Freeswitch will not know this until advised by the B > leg UA via a 415 Bad Security Level responce from the B leg INVITE. > > Per debug attached to original email, FS appears to generate the SRTP > descriptors however does not add them to the second INVITE sent to the > B leg. ?IMHO this is a fault and should be corrected. ?Anthony, do you > have any thoughts on this! > > Call testing shows the following results. > 1. ?A leg INVITE with SRTP descriptors in SDP and sip_secure_media set > in the dialplan. ?B leg INVITE has no SRTP descriptors in SDP . ?RTP > between A UA and FS uses SRTP, B leg does not. > > 2. ?A leg INVITE with SRTP descriptors in SDP and sip_secure_media and > export sip_secure_media=true set in the dialplan. ?B leg INVITE also > SRTP descriptors in SDP . ?RTP between A UA and FS uses SRTP, FS and B > UA also uses SRTP. > > 3. ?A leg INVITE with no SRTP descriptors in SDP and export > sip_secure_media=true set in the dialplan. ?B leg INVITE has SRTP > descriptors in SDP. ?RTP between B UA and FS uses SRTP, A leg does > not. > > 4. ?A leg INVITE without SRTP descriptors in SDP, B leg INVITE without > SRTP descriptors in SDP results in 415 Bad Security Level. ?Dialplan > set to continue on fail and export sip_secure_media=true then bridge > the call once more. ?Debug shows that FS generates the SRTP > descriptors, however FS does not add them to the second INVITE. > > As you can see from above, FS can set SRTP on a per leg basis. > However for some reason it fails to add the SRTP descriptors to the > SDP in the second INVITE for scenario 3. > > I hope this has cleared up the confusion regarding my original email. > If you wish to discuss further please let me know what time the > conference is and I can join in. > > Regards, > Jim > > > > > On Fri, May 22, 2009 at 11:59 PM, Brian West wrote: >> >> On May 22, 2009, at 12:47 AM, Jim Burke wrote: >> >> Hey Brian, >> >> Will have a look at ZRTP :) >> >> Not sure I understand your comments regarding its all over once >> receiving the 415 from the B party. ?Is'nt that what parm >> continue_on_fail does? ?The fact that it sends the invite back out >> sorta proves this. >> >> The A-LEG has to hangup to re-enable SRTP it can't do it if it didn't invite >> with it in the first place. >> >> The other point of interest here is that if you set > application="export" data="sip_secure_media=true"/> before the first >> bridge function it will include the security descriptions in the B leg >> INVITE even when the A leg does not have them and the call will >> succeed. ?The B Eyebeam will show the locked padlock while A does not. >> >> Make sure you do not answer the call before you do it. >> >> From what I can see in code it is this guy that must stop it all from >> >> happening. ?TFLAG_SECURE ?But I dont understand why :( >> >> Again you have to invite to FS with crypto it can't magically cause crypto >> to work unless you initiate it with your first invite. >> >> Regards, >> Jim >> >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! ?http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From nicholsonster at gmail.com Mon May 25 23:05:28 2009 From: nicholsonster at gmail.com (John Nicholson) Date: Tue, 26 May 2009 01:05:28 -0500 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <4A1B2D26.90202@gmx.net> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> Message-ID: <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> Virtualization has issues with timing in my experiance. Sent from my iPhone On May 25, 2009, at 6:43 PM, Peter P GMX wrote: > We have used FS on ec2 for testing purposes only. It was ok. We havn't > done any performance test though. > > Best regards > Peter > > Ing. Edwin Villarreal schrieb: >> >> Hello my friends. >> >> >> >> Has anyone used the EC2 for production? Tests? >> >> >> >> I?m wondering if it would be ?better? to have a FS system in >> the cloud >> for carrier-to-carrier connections. >> >> >> >> Any ideas will be appreciated >> >> >> >> Thanks 2 u all >> >> >> >> *Edwin Villarreal* >> >> World Net Commerce SA CV >> >> WNC Telecom >> >> >> >> --- >> --------------------------------------------------------------------- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Mon May 25 23:49:48 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 26 May 2009 16:49:48 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> Message-ID: <20090526064948.GA26484@jdc.jasonjgw.net> Jim Burke wrote: > > > > Why are you setting the same variable twice? The result will be that only the second of the above set commands takes effect, so you should delete the first of them. From jim at evolutiontel.net Tue May 26 00:33:20 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 26 May 2009 17:33:20 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: <20090526064948.GA26484@jdc.jasonjgw.net> References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> <20090526064948.GA26484@jdc.jasonjgw.net> Message-ID: If I understand your comment correctly, I did not have both of the above snippets in the dialplan at the same time. The dialplan was modified continually to get the correct vars that worked for my situation and then reloadxml to get them working. Regards, On Tue, May 26, 2009 at 4:49 PM, Jason White wrote: > Jim Burke wrote: > >> ? ? >> ? ? ? >> ? ? ? ? >> ? ? ? ? > > Why are you setting the same variable twice? The result will be that only the > second of the above set commands takes effect, so you should delete the first > of them. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Tue May 26 01:09:50 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 26 May 2009 18:09:50 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> <20090526064948.GA26484@jdc.jasonjgw.net> Message-ID: <20090526080950.GA30893@jdc.jasonjgw.net> Jim Burke wrote: > If I understand your comment correctly, I did not have both of the > above snippets in the dialplan at the same time. The dialplan was > modified continually to get the correct vars that worked for my > situation and then reloadxml to get them working. Right, you can't have one after the other like this, because it is equivalent to since you're overriding the setting of the variable in the very next line. From jim at evolutiontel.net Tue May 26 02:11:29 2009 From: jim at evolutiontel.net (Jim Burke) Date: Tue, 26 May 2009 19:11:29 +1000 Subject: [Freeswitch-users] Secure RTP In-Reply-To: <20090526080950.GA30893@jdc.jasonjgw.net> References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> <20090526064948.GA26484@jdc.jasonjgw.net> <20090526080950.GA30893@jdc.jasonjgw.net> Message-ID: Ahh right now I get what you are saying, I thought from the wiki that I would have to set the feature on and then tell it what cause codes I wanted to trap. Will fix up my dialplan cause I don't want it to trap other causes for this scenario. Thanks! On Tue, May 26, 2009 at 6:09 PM, Jason White wrote: > Jim Burke wrote: >> If I understand your comment correctly, I did not have both of the >> above snippets in the dialplan at the same time. ?The dialplan was >> modified continually to get the correct vars that worked for my >> situation and then reloadxml to get them working. > > Right, you can't have > > > one after the other like this, because it is equivalent to > > since you're overriding the setting of the variable in the very next line. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pw at le7.no Tue May 26 02:45:41 2009 From: pw at le7.no (IPW) Date: Tue, 26 May 2009 11:45:41 +0200 Subject: [Freeswitch-users] err after installling freeswitch In-Reply-To: References: <557185.81756.qm@web34302.mail.mud.yahoo.com> <20090525021737.GA31033@jdc.jasonjgw.net> <4A1A1DCC.1050201@telkom.co.id> Message-ID: <001001c9dde6$bb7874e0$32695ea0$@no> Hei! Hvilken spedit?r? Per Westr?m Ing. Per Westr?m, Leiv Eirikssonsgt. 7, 0271 Oslo, Norway Tlf. + 47 22444550, Fax + 47 22554964, Mail pw at le7.no [mailto:pw at le7.no -----Opprinnelig melding----- Fra: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne av Michael Jerris Sendt: 26. mai 2009 03:18 Til: freeswitch-users at lists.freeswitch.org Emne: Re: [Freeswitch-users] err after installling freeswitch This looks like a permissions error on creating or opening files in the db directory. Mike On May 25, 2009, at 12:25 AM, mashudi wrote: > Hi Guys, > I have install Freeswitch with version : FreeSWITCH Version 1.0.4pre7 > (13238M) Started. > I load the openzap module after install the wanpipe modul and > everything > running , after I query the status of the Sangoma Card (A104D quad E1) > we got error like this : > 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 > switch_core_db_persistant_execute_trans() SQL ERR [unable to open > database file] > 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 > switch_core_db_persistant_execute_trans() SQL ERR [unable to open > database file] > 2009-05-25 11:07:45 [ERR] switch_core_sqldb.c:95 > switch_core_db_persistant_execute_trans() SQL ERR [unable to open > database file] > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brad.tuan at gmail.com Tue May 26 03:43:47 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 26 May 2009 18:43:47 +0800 Subject: [Freeswitch-users] 407 Proxy Authentication Required In-Reply-To: <20090526014012.GA16268@jdc.jasonjgw.net> References: <20090526014012.GA16268@jdc.jasonjgw.net> Message-ID: I am sure the Auth-call is closed and the acl is closed. And i didn't change any setting. But,the response message change from "407 Proxy Authentication Required" to "480 Temporarily Unavailable" today......... Anybody can tell me what happen?? 2009-05-26 17:37:50 [INFO] sofia_presence.c:617 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2009-05-26 17:37:50 [WARNING] sofia_presence.c:571 actual_sofia_presence_event_handler() external is passive, skipping 2009-05-26 17:37:50 [NOTICE] switch_core_session.c:1020 switch_core_session_thread() Close Channel sofia/internal/97719006 at 163.28.2.7:5060 [CS_DONE] 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() 203.64.215.209 is an alias, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() default is an alias, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() nat is an alias, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() outbound is an alias, skipping 2009-05-26 17:37:50 [INFO] sofia_presence.c:601 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal) 2009-05-26 17:37:50 [ERR] sofia_presence.c:609 actual_sofia_presence_event_handler() DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Call Ended','unknown','163.28.2.7',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='97719006' and (sub_to_host='163.28.2.7' or presence_hosts like '%163.28.2.7%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [0cfc45ae-5cb5-0148-aa9b-3ed9fb0f8255] FreeSWITCH-Hostname: [xpp-8435d6cb386] FreeSWITCH-IPv4: [203.64.215.209] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-05-26 17:37:50] Event-Date-GMT: [Tue, 26 May 2009 09:37:50 GMT] Event-Date-Timestamp: [1243330670130868] Event-Calling-File: [switch_channel.c] Event-Calling-Function: [switch_channel_presence] Event-Calling-Line-Number: [467] Channel-State: [CS_HANGUP] Channel-State-Number: [10] Channel-Name: [sofia/internal/97719006 at 163.28.2.7:5060] Unique-ID: [9218a3e4-04be-1d4a-97ea-8bdcb20ae3b7] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Caller-Username: [97719006] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [97719006] Caller-Caller-ID-Number: [97719006] Caller-Network-Addr: [163.28.32.51] Caller-Destination-Number: [97730009] Caller-Unique-ID: [9218a3e4-04be-1d4a-97ea-8bdcb20ae3b7] Caller-Source: [mod_sofia] Caller-Context: [public] Caller-Channel-Name: [sofia/internal/97719006 at 163.28.2.7:5060] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1243330670083993] Caller-Channel-Created-Time: [1243330670083993] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [1243330670130868] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] proto: [..\..\src\switch_channel.c] login: [..\..\src\switch_channel.c] from: [97719006 at 163.28.2.7] rpid: [unknown] status: [CS_HANGUP] event_type: [presence] alt_event_type: [dialog] event_count: [3] 2009-05-26 17:37:50 [INFO] sofia_presence.c:617 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2009-05-26 17:37:50 [WARNING] sofia_presence.c:571 actual_sofia_presence_event_handler() external is passive, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() 203.64.215.209 is an alias, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() default is an alias, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() nat is an alias, skipping 2009-05-26 17:37:50 [WARNING] sofia_presence.c:564 actual_sofia_presence_event_handler() outbound is an alias, skipping 2009-05-26 17:37:50 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/internal/97719006 at 163.28.2.7:5060[cc379275-1164-d040-b777-f4b8b3b542fb] 2009-05-26 17:37:50 [DEBUG] sofia.c:2806 sofia_handle_sip_i_state() Channel sofia/internal/97719006 at 163.28.2.7:5060 entering state [received] 2009-05-26 17:37:50 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=msx1 0 0 IN IP4 163.28.2.7 s=sip call c=IN IP4 61.61.162.130 t=0 0 m=audio 5064 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 2009-05-26 17:37:50 [DEBUG] sofia_glue.c:2606 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-05-26 17:37:50 [DEBUG] sofia_glue.c:2606 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:390 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) Running State Change CS_NEW 2009-05-26 17:37:50 [DEBUG] sofia_glue.c:1741 sofia_glue_tech_set_codec() Set Codec sofia/internal/97719006 at 163.28.2.7:5060 PCMU/8000 20 ms 160 samples 2009-05-26 17:37:50 [DEBUG] sofia_glue.c:2566 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-26 17:37:50 [DEBUG] sofia.c:2962 sofia_handle_sip_i_state() ( sofia/internal/97719006 at 163.28.2.7:5060) State Change CS_NEW -> CS_INIT 2009-05-26 17:37:50 [DEBUG] switch_core_session.c:868 switch_core_session_signal_state_change() Send signal sofia/internal/97719006 at 163.28.2.7:5060 [BREAK] 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:396 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State NEW 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:390 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) Running State Change CS_INIT 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:469 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State INIT 2009-05-26 17:37:50 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/97719006 at 163.28.2.7:5060 SOFIA INIT 2009-05-26 17:37:50 [DEBUG] mod_sofia.c:111 sofia_on_init() ( sofia/internal/97719006 at 163.28.2.7:5060) State Change CS_INIT -> CS_ROUTING 2009-05-26 17:37:50 [DEBUG] switch_core_session.c:868 switch_core_session_signal_state_change() Send signal sofia/internal/97719006 at 163.28.2.7:5060 [BREAK] 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:469 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State INIT going to sleep 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:390 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) Running State Change CS_ROUTING 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:472 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State ROUTING 2009-05-26 17:37:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/97719006 at 163.28.2.7:5060 SOFIA ROUTING 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:71 switch_core_standard_on_routing() sofia/internal/97719006 at 163.28.2.7:5060Standard ROUTING 2009-05-26 17:37:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 97719006->97730009 in context public Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 parsing [public->unloop] continue=false Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 parsing [public->outside_call] continue=true Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Absolute Condition [outside_call] Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Action set(outside_call=true) Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 parsing [public->call_debug] continue=true Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 parsing [public->public_extensions] continue=false Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Regex (FAIL) [public_extensions] destination_number(97730009) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 parsing [public->public_did] continue=false Dialplan: sofia/internal/97719006 at 163.28.2.7:5060 Regex (FAIL) [public_did] destination_number(97730009) =~ /^(5551212)$/ break=on-false 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:107 switch_core_standard_on_routing() (sofia/internal/97719006 at 163.28.2.7:5060) State Change CS_ROUTING -> CS_EXECUTE 2009-05-26 17:37:50 [DEBUG] switch_core_session.c:868 switch_core_session_signal_state_change() Send signal sofia/internal/97719006 at 163.28.2.7:5060 [BREAK] 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:472 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State ROUTING going to sleep 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:390 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) Running State Change CS_EXECUTE 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:479 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State EXECUTE 2009-05-26 17:37:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/97719006 at 163.28.2.7:5060 SOFIA EXECUTE 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/internal/97719006 at 163.28.2.7:5060Standard EXECUTE EXECUTE sofia/internal/97719006 at 163.28.2.7:5060 set(outside_call=true) 2009-05-26 17:37:50 [DEBUG] mod_dptools.c:747 set_function() sofia/internal/97719006 at 163.28.2.7:5060 SET [outside_call]=[true] 2009-05-26 17:37:50 [NOTICE] switch_core_state_machine.c:172 switch_core_standard_on_execute() Hangup sofia/internal/97719006 at 163.28.2.7:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-26 17:37:50 [DEBUG] switch_channel.c:1612 switch_channel_perform_hangup() Send signal sofia/internal/97719006 at 163.28.2.7:5060 [KILL] 2009-05-26 17:37:50 [DEBUG] switch_core_session.c:868 switch_core_session_signal_state_change() Send signal sofia/internal/97719006 at 163.28.2.7:5060 [BREAK] 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:479 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State EXECUTE going to sleep 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:390 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) Running State Change CS_HANGUP 2009-05-26 17:37:50 [DEBUG] switch_core_state_machine.c:430 switch_core_session_run() (sofia/internal/97719006 at 163.28.2.7:5060) State HANGUP 2009-05-26 17:37:50 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/97719006 at 163.28.2.7:5060 hanging up, cause: NORMAL_CLEARING 2009-05-26 17:37:50 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480 2009/5/26 Jason White > Brad Tuan wrote: > > So , Anyone please tell me how can i do?? > > Turning off the ACL and setting auth-calls to false should be enough to do > it. > > To find out where the problem is in your configuration, set the log level > to > debug if it isn't already, and read the logs carefully. > > You can also turn on SIP tracing if necessary. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/269a8096/attachment-0001.html From jason at jasonjgw.net Tue May 26 04:07:30 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 26 May 2009 21:07:30 +1000 Subject: [Freeswitch-users] 407 Proxy Authentication Required In-Reply-To: References: <20090526014012.GA16268@jdc.jasonjgw.net> Message-ID: <20090526110730.GA8677@jdc.jasonjgw.net> Brad Tuan wrote: > But,the response message change from "407 Proxy Authentication Required" to > "480 Temporarily Unavailable" today......... > > Anybody can tell me what happen?? Your SIP trace might give you a clue as to what happened. sofia profile external siptrace on (substituting the relevant profile for "external" in the above command, as required, and repeating the command if more than one profile is involved). From anthony.minessale at gmail.com Tue May 26 05:58:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 May 2009 07:58:25 -0500 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> Message-ID: <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> This one happens to every new guy trying to make FS into a dialer app using JS. for every sessionX you create in js with the new Session constructor sessionX.setAutoHangup(0); This allows the channels to remain alive outside the script once they are hungup/transferred etc. On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: > Hi, > I had some discussion with the IRC regarding the uuid_transfer gets > hang-up where the call is originated via javascript thru event socket. I was > suggested to install latest SVN trunk. I did that again the same issue, the > log is attached with here http://pastebin.freeswitch.org/9103 > > My call flow like this, > > 1. api jsrun fils.js > 2. capture the uuid > 3. api uuid_transfer -both > > Both the leg gets hangedup. > > Someone can assist me where I am wrong. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/eb0ae4e7/attachment.html From niall.crosby at gmail.com Tue May 26 07:10:00 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Tue, 26 May 2009 15:10:00 +0100 Subject: [Freeswitch-users] Echo cancellation needed? Message-ID: <4aec92830905260710u18156d6fh21a5677b6042f97b@mail.gmail.com> Dear FS users, If I am getting Sangoma hardware to connect Freeswitch to E1, should I get a card with echo cancellation or not? Also, is there any advantage to PCIe over PCI? (I understand the difference between these and am gonig with PCIe, just checking encase there are legacy issues that make PCIe wrong choice) Many thanks, Niall. From brian at freeswitch.org Tue May 26 07:27:57 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 09:27:57 -0500 Subject: [Freeswitch-users] 407 Proxy Authentication Required In-Reply-To: <20090526110730.GA8677@jdc.jasonjgw.net> References: <20090526014012.GA16268@jdc.jasonjgw.net> <20090526110730.GA8677@jdc.jasonjgw.net> Message-ID: <3A90897C-C57D-4F52-8E98-EA605A3A3090@freeswitch.org> Looks like his outbound call is failing now. /b On May 26, 2009, at 6:07 AM, Jason White wrote: > Your SIP trace might give you a clue as to what happened. > sofia profile external siptrace on > (substituting the relevant profile for "external" in the above > command, as > required, and repeating the command if more than one profile is > involved). From brian at freeswitch.org Tue May 26 07:29:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 09:29:44 -0500 Subject: [Freeswitch-users] Secure RTP In-Reply-To: References: <78E6F382-B8E0-4BA8-8545-9691CE30F580@freeswitch.org> Message-ID: I know this but you don't have to have it in x-lite or eyebeam directly. You just need the zfone application along with Eyebeam or X- Lite right now. /b On May 26, 2009, at 12:52 AM, Jim Burke wrote: > > FYI...According to Counterpath ZRTP is not added to the retail > versions of Eyebeam or X-Lite, you need to purchase a bulk order. From brian at freeswitch.org Tue May 26 07:31:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 09:31:32 -0500 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> Message-ID: Not with FreeSWITCH in our testing. Now if you have stupid defaults in your virtualization env. it might act funny but I have run FS on EC2 without a problem. /b On May 26, 2009, at 1:05 AM, John Nicholson wrote: > Virtualization has issues with timing in my experiance. > > Sent from my iPhone From brian at freeswitch.org Tue May 26 07:32:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 09:32:42 -0500 Subject: [Freeswitch-users] how to disbale : switch_core_sqldb() In-Reply-To: <4A1B5A96.8060003@telkom.co.id> References: <4A1A47BC.1020905@telkom.co.id> <4A1B5A96.8060003@telkom.co.id> Message-ID: <5572BDB9-0D87-45E0-93AD-6CB71909AAAB@freeswitch.org> I didn't see an error line maybe you have permission problems butyou just pasted only the INFO line. You do know if you disable the sql you loose the ability to show calls and channels? /b On May 25, 2009, at 9:57 PM, mashudi wrote: > Dear Muhammad Shahzad , > it is work and the err after has gone, thank you very much for your > suggestion. From brian at freeswitch.org Tue May 26 07:35:41 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 09:35:41 -0500 Subject: [Freeswitch-users] Echo cancellation needed? In-Reply-To: <4aec92830905260710u18156d6fh21a5677b6042f97b@mail.gmail.com> References: <4aec92830905260710u18156d6fh21a5677b6042f97b@mail.gmail.com> Message-ID: <859FFFE5-1DE3-4EC4-8EFB-4815612D0E18@freeswitch.org> On May 26, 2009, at 9:10 AM, Niall Crosby wrote: > Dear FS users, > > If I am getting Sangoma hardware to connect Freeswitch to E1, should I > get a card with echo cancellation or not? I would use hardware echo cancel if at all possible. > Also, is there any advantage to PCIe over PCI? (I understand the > difference between these and am gonig with PCIe, just checking encase > there are legacy issues that make PCIe wrong choice) If its just one E1 it won't matter.. > > > Many thanks, > Niall. From plite2012 at gmail.com Mon May 25 22:03:42 2009 From: plite2012 at gmail.com (Paul Li) Date: Tue, 26 May 2009 00:03:42 -0500 Subject: [Freeswitch-users] IVR xml dialplan for outbound calls Message-ID: There are a few examples in the wiki, showing how to configure IVR for inbound calls. My question lies in whether it is possible to write a dialplan in xml or scripts to configure IVR for outbound calls. Here is a typical scenario (of usage): (1) Call any extension or external endpoints (maybe PSTN numbers); (2) Activate IVR menu once the recipient answers call; (3) Take feedback (DTMF keystrokes or survey recordings). Your assistance is highly appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/3b166c26/attachment-0001.html From ryder86 at googlemail.com Tue May 26 01:27:16 2009 From: ryder86 at googlemail.com (=?KOI8-R?B?4dLUxc0g98HTyczYxdc=?=) Date: Tue, 26 May 2009 12:27:16 +0400 Subject: [Freeswitch-users] Unicast isn't working Message-ID: Hi, I am trying to setup ASR in FreeSwitch using Nuance ASR server and MRCP. Both FreeSwitch and Nuance installed on Windows Server 2003. FreeSwitch version is 1.0.3 (12567M) I found an example in Perl at http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl and decided to do the same in C#. It establishes connection with Nuance and loads the grammar, everything works fine. The next step is to capture audio from FS and tramsmit it to ASR. This can be done with unicast. We must create an outbound socket and issue "unicast" command. Here it goes: private void SetupAudioTransmission() { EventSocket = new Socket(AddressFamily.InterNetwork, SocketType.Stream, ProtocolType.Tcp); EventSocket.SendTimeout = Config.Timeout; int ESPort = SearchESPort(EventSocket); Thread thrESListener = new Thread(new ThreadStart(ListenerThreadStart)); thrESListener.Start(); WriteLog(LogLevel.Info, "Creating outbound event socket"); Session.Execute("socket", "127.0.0.1:" + ESPort); //main thread stops, listener thread listens for outbound socket connection. WriteLog(LogLevel.Info, "Outbound event socket disconnected"); EventSocket.Close(); } //here we accept outbound socket and transmit unicast command through it private void ListenerThreadStart() { Socket sockHandler = EventSocket.Accept(); WriteLog(LogLevel.Info, "Incoming connection"); sockHandler.Send(MessageEncoding.GetBytes("Connect\n\n")); WriteLog(LogLevel.Info, GetServerResponse(sockHandler)); int rtpPort = (RTPSocket.RemoteEndPoint as IPEndPoint).Port; string command = string.Format("sendmsg\r\ncall-command: unicast\r\nlocal-ip: {0}\r\nlocal-port: {1}\r\nremote-ip: {2}\r\n" + "remote-port: {3}\r\ntransport: udp\r\nflags: native\r\n\r\n", Config.LocalIP, rtpPort + 1, Config.LocalIP, rtpPort); WriteLog(LogLevel.Info, command); sockHandler.Send(MessageEncoding.GetBytes(command)); WriteLog(LogLevel.Info, GetServerResponse(sockHandler)); sockHandler.Disconnect(false); sockHandler.Close(); } After this, FS writes that unicast has been created on corresponding IPs and ports. It really creates an UDP socket, but doesn't transmit any data through it. I tested it with Wireshark and from my application, nothing was detected. Also, if we specify "transport:tcp" in unicast command, it uses UDP anyway, that's strange. Here is how I listen UDP packets. private void DetectSpeech() { WriteLog(LogLevel.Info, "Reading audio"); byte[] FSRecvBuf = new byte[2048]; IPEndPoint epFS = new IPEndPoint(IPAddress.Loopback, (RTPSocket.RemoteEndPoint as IPEndPoint).Port); FSSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, ProtocolType.Udp); FSSocket.ReceiveTimeout = 100000; FSSocket.SendTimeout = 100000; WriteLog(LogLevel.Info, "Binding socket to " + epFS.Port); FSSocket.Bind(epFS); FSSocket.Connect(new IPEndPoint(IPAddress.Loopback, epFS.Port + 1)); while (Session.Ready()) { int recvCount = FSSocket.Receive(FSRecvBuf); WriteLog(LogLevel.Info, "Received bytes: " + recvCount); } } Can someone help me to solve this? Do I do something wrong or I forgot something or it doesn't work at all? Best regards. Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/d39ff9e8/attachment-0001.html From jcromes at gmail.com Mon May 25 19:35:52 2009 From: jcromes at gmail.com (j3flight) Date: Mon, 25 May 2009 19:35:52 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? Message-ID: <23715721.post@talk.nabble.com> I'm attempting to replicate the behavior of an Asterisk system with FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to be supported (easily). Ok, so I've setup my dialplan so that when a specific extension is hit, it calls out to some javascript which acts like an IVR to handle the conference setup. (Similar to this: http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own improvements.) Anyway, the conferences are stored permanently in a database, but I want them protected by their "owner" so they can only be used when that conference owner dials. If other users have entered the conference prior to the owner, they should hear music-on-hold until the leader enters. This is easy in Asterisk because you can pop someone into MeetMe with different flags. So, in my IVR, I prompt for the "conference number" (known to all) and then the "password" (known only to the owner/leader). If the proper password is entered, the user is sent to conference XYZ with the leader flag set. If no password is entered, the user goes to conference XYZ, without the leader flag. If anyone enters before the leader, they're told by MeetMe that "the conference will begin when the leader arrives" and MeetMe provides MOH until that time. Help! This is an absolute deal-breaker for my install... How can I do this in FreeSwitch? Thanks... -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23715721p23715721.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jcromes at gmail.com Tue May 26 06:20:54 2009 From: jcromes at gmail.com (j3flight) Date: Tue, 26 May 2009 06:20:54 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? Message-ID: <23715721.post@talk.nabble.com> I'm attempting to replicate the behavior of an Asterisk system with FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to be supported (easily). Ok, so I've setup my dialplan so that when a specific extension is hit, it calls out to some javascript which acts like an IVR to handle the conference setup. (Similar to this: http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own improvements.) Anyway, the conferences are stored permanently in a database, but I want them protected by their "owner" so they can only be used when that conference owner dials. If other users have entered the conference prior to the owner, they should hear music-on-hold until the leader enters. This is easy in Asterisk because you can pop someone into MeetMe with different flags. So, in my IVR, I prompt for the "conference number" (known to all) and then the "password" (known only to the owner/leader). If the proper password is entered, the user is sent to conference XYZ with the leader flag set. If no password is entered, the user goes to conference XYZ, without the leader flag. If anyone enters before the leader, they're told by MeetMe that "the conference will begin when the leader arrives" and MeetMe provides MOH until that time. Help! This is an absolute deal-breaker for my install... How can I do this in FreeSwitch? Thanks... -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23715721p23715721.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jcromes at gmail.com Tue May 26 07:38:06 2009 From: jcromes at gmail.com (Jason Cromes) Date: Tue, 26 May 2009 09:38:06 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? Message-ID: <4A1BFECE.7070603@gmail.com> First off, I apologize if this has been sent multiple times, the mailing list won't cooperate with me... Hopefully that is resolved now. I'm attempting to replicate the behavior of an Asterisk conferencing system and I need a feature that, I'm surprised to say, doesn't seem to be supported (easily) in FreeSwitch. I've setup my dialplan so that when a specific extension is hit, it calls out to some javascript which acts like an IVR to handle the prompts and conferencing setup. (Similar to this: http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own improvements.) Anyway, the conferences are stored permanently in a database table, but I need them disabled until the conference owner dials in and enables his conference. If other users have entered the conference prior to the owner, they should hear music-on-hold until the leader enters. (The purpose of this is to prevent abuse of conferences that are, basically, always available.) This is easy in Asterisk because you can pop someone into MeetMe with different flags. So, in my IVR, I prompt for the "conference number" (known to all) and then the "password" (known only to the owner/leader). If the proper password is entered, that user is sent to conference XYZ with the leader flag set. If no password is entered, the user goes to conference XYZ, without the leader flag. If anyone enters before the leader, they're told by MeetMe that "the conference will begin when the leader arrives" and MeetMe provides MOH until that time. Help! This is an absolute deal-breaker for my install... How can I do this in FreeSwitch? Thanks... From niall.crosby at gmail.com Tue May 26 07:41:55 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Tue, 26 May 2009 15:41:55 +0100 Subject: [Freeswitch-users] Echo cancellation needed? In-Reply-To: <859FFFE5-1DE3-4EC4-8EFB-4815612D0E18@freeswitch.org> References: <4aec92830905260710u18156d6fh21a5677b6042f97b@mail.gmail.com> <859FFFE5-1DE3-4EC4-8EFB-4815612D0E18@freeswitch.org> Message-ID: <4aec92830905260741v50b396bvec0e8a4b337e1213@mail.gmail.com> Thanks Brian. 2009/5/26 Brian West : > > On May 26, 2009, at 9:10 AM, Niall Crosby wrote: > >> Dear FS users, >> >> If I am getting Sangoma hardware to connect Freeswitch to E1, should I >> get a card with echo cancellation or not? > > I would use hardware echo cancel if at all possible. > >> Also, is there any advantage to PCIe over PCI? (I understand the >> difference between these and am gonig with PCIe, just checking encase >> there are legacy issues that make PCIe wrong choice) > > If its just one E1 it won't matter.. > >> >> >> Many thanks, >> Niall. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -- Sremium Ltd. Reg Number: 451937 Mobile: +353 (0)87 2393174 Web: www.sremium.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of Sremium. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. From brian at freeswitch.org Tue May 26 07:51:00 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 09:51:00 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A1BFECE.7070603@gmail.com> References: <4A1BFECE.7070603@gmail.com> Message-ID: <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> Jason, There are many ways to accomplish this using FreeSWITCH. All of which will require you to do a little bit of coding in js, lua or some other language. 1. Park all callers into a fifo.. (see mod_fifo) 2. When leader auths in your script then you uuid_transfer them all into the conference. You could also do this via esl using perl, lua, python, php or ruby. That should accomplish the same ending result. /b On May 26, 2009, at 9:38 AM, Jason Cromes wrote: > This is easy in Asterisk because you can pop someone into MeetMe with > different flags. So, in my IVR, I prompt for the "conference number" > (known to all) and then the "password" (known only to the owner/ > leader). > If the proper password is entered, that user is sent to conference XYZ > with the leader flag set. If no password is entered, the user goes to > conference XYZ, without the leader flag. If anyone enters before the > leader, they're told by MeetMe that "the conference will begin when > the > leader arrives" and MeetMe provides MOH until that time. > > Help! This is an absolute deal-breaker for my install... How can I > do > this in FreeSwitch? > Thanks... From kristian.kielhofner at gmail.com Tue May 26 08:11:17 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 26 May 2009 11:11:17 -0400 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> Message-ID: <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> On Tue, May 26, 2009 at 10:31 AM, Brian West wrote: > Not with FreeSWITCH in our testing. ?Now if you have stupid defaults > in your virtualization env. it might act funny but I have run FS on > EC2 without a problem. > > /b Hey Brian, FreeSWITCH in EC2 is a bit of a mystery to me... Call me old fashioned but in my mind VoIP and geography are linked in %99 of scenarios. Having VoIP services in a pure "cloud" environment just doesn't sound like a good idea to me. Consider a "hosted" environment with clients registered to a FreeSWITCH server. One day your instance is physically running on hardware in Seattle. The next day it could (potentially) be running in Chicago. That's obviously a very different routing path for your clients. Even /if/ Amazon (or whomever) employs every routing/network trick in the book you still won't be able to get over that change in geography. It's certainly possible a change like this may very well never happen in practice. I wouldn't know; I've never used EC2 and I don't even know that much about it. I'm just curious how well strictly, practically speaking this will work in the long term. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Tue May 26 08:20:09 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 10:20:09 -0500 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> Message-ID: On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote: > Hey Brian, > > FreeSWITCH in EC2 is a bit of a mystery to me... > > Call me old fashioned but in my mind VoIP and geography are linked > in %99 of scenarios. Having VoIP services in a pure "cloud" > environment just doesn't sound like a good idea to me. > > Consider a "hosted" environment with clients registered to a > FreeSWITCH server. One day your instance is physically running on > hardware in Seattle. The next day it could (potentially) be running > in Chicago. That's obviously a very different routing path for your > clients. Even /if/ Amazon (or whomever) employs every routing/network > trick in the book you still won't be able to get over that change in > geography. For some people this isn't a huge difference... now if it were to swap continents then yes it would be a problem. But I haven't seen Amazon do this but I haven't left the instances up long enough to see. > It's certainly possible a change like this may very well never > happen in practice. I wouldn't know; I've never used EC2 and I don't > even know that much about it. I'm just curious how well strictly, > practically speaking this will work in the long term. There are other companies that do this stuff but personally me... I want my stuff running on real hardware. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/8ce03a11/attachment.html From dujinfang at gmail.com Tue May 26 08:22:55 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 26 May 2009 23:22:55 +0800 Subject: [Freeswitch-users] how to enable ESL for ruby? Message-ID: Hi, Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library On MacOSX 10.5, I can't get ESL for ruby work. make throws error: sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make rubymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/Users/ seven/workspace/test/freeswitch/trunk/libs/esl/src/include - DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb - I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C ruby g++ -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c esl_wrap.cpp -o esl_wrap.o g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. Undefined symbols: "_main", referenced from: start in crt1.10.5.o ld: symbol(s) not found collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 Thanks for any help. From dave at 3c.co.uk Tue May 26 08:08:55 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 26 May 2009 16:08:55 +0100 Subject: [Freeswitch-users] Unicast isn't working References: Message-ID: Hi Artem, Please to see that some of the stuff I wrote is useful to someone..! I've written an FS module which will send the audio over - it's more efficient than using unicast. Let me know if you'd like a copy. Cheers -- Dave ----- Original Message ----- From: ????? ???????? To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, May 26, 2009 9:27 AM Subject: [Freeswitch-users] Unicast isn't working Hi, I am trying to setup ASR in FreeSwitch using Nuance ASR server and MRCP. Both FreeSwitch and Nuance installed on Windows Server 2003. FreeSwitch version is 1.0.3 (12567M) I found an example in Perl at http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl and decided to do the same in C#. It establishes connection with Nuance and loads the grammar, everything works fine. The next step is to capture audio from FS and tramsmit it to ASR. This can be done with unicast. We must create an outbound socket and issue "unicast" command. Here it goes: private void SetupAudioTransmission() { EventSocket = new Socket(AddressFamily.InterNetwork, SocketType.Stream, ProtocolType.Tcp); EventSocket.SendTimeout = Config.Timeout; int ESPort = SearchESPort(EventSocket); Thread thrESListener = new Thread(new ThreadStart(ListenerThreadStart)); thrESListener.Start(); WriteLog(LogLevel.Info, "Creating outbound event socket"); Session.Execute("socket", "127.0.0.1:" + ESPort); //main thread stops, listener thread listens for outbound socket connection. WriteLog(LogLevel.Info, "Outbound event socket disconnected"); EventSocket.Close(); } //here we accept outbound socket and transmit unicast command through it private void ListenerThreadStart() { Socket sockHandler = EventSocket.Accept(); WriteLog(LogLevel.Info, "Incoming connection"); sockHandler.Send(MessageEncoding.GetBytes("Connect\n\n")); WriteLog(LogLevel.Info, GetServerResponse(sockHandler)); int rtpPort = (RTPSocket.RemoteEndPoint as IPEndPoint).Port; string command = string.Format("sendmsg\r\ncall-command: unicast\r\nlocal-ip: {0}\r\nlocal-port: {1}\r\nremote-ip: {2}\r\n" + "remote-port: {3}\r\ntransport: udp\r\nflags: native\r\n\r\n", Config.LocalIP, rtpPort + 1, Config.LocalIP, rtpPort); WriteLog(LogLevel.Info, command); sockHandler.Send(MessageEncoding.GetBytes(command)); WriteLog(LogLevel.Info, GetServerResponse(sockHandler)); sockHandler.Disconnect(false); sockHandler.Close(); } After this, FS writes that unicast has been created on corresponding IPs and ports. It really creates an UDP socket, but doesn't transmit any data through it. I tested it with Wireshark and from my application, nothing was detected. Also, if we specify "transport:tcp" in unicast command, it uses UDP anyway, that's strange. Here is how I listen UDP packets. private void DetectSpeech() { WriteLog(LogLevel.Info, "Reading audio"); byte[] FSRecvBuf = new byte[2048]; IPEndPoint epFS = new IPEndPoint(IPAddress.Loopback, (RTPSocket.RemoteEndPoint as IPEndPoint).Port); FSSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, ProtocolType.Udp); FSSocket.ReceiveTimeout = 100000; FSSocket.SendTimeout = 100000; WriteLog(LogLevel.Info, "Binding socket to " + epFS.Port); FSSocket.Bind(epFS); FSSocket.Connect(new IPEndPoint(IPAddress.Loopback, epFS.Port + 1)); while (Session.Ready()) { int recvCount = FSSocket.Receive(FSRecvBuf); WriteLog(LogLevel.Info, "Received bytes: " + recvCount); } } Can someone help me to solve this? Do I do something wrong or I forgot something or it doesn't work at all? Best regards. Artem ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/2094b41b/attachment.html From brian at freeswitch.org Tue May 26 08:27:38 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 10:27:38 -0500 Subject: [Freeswitch-users] how to enable ESL for ruby? In-Reply-To: References: Message-ID: <9D8ADB1E-123A-4347-8A31-B13BE3C760D3@freeswitch.org> The makefile will have to be changed to work with OS X since the linking is done differently. It would be very similar to this one http://www.bkw.org/esl.imac.diff Below will get it to compile: imac:esl brian$ svn diff Index: ruby/Makefile =================================================================== --- ruby/Makefile (revision 13432) +++ ruby/Makefile (working copy) @@ -1,6 +1,7 @@ ## no idea how to simply ask ruby which one to use LOCAL_CFLAGS=-I$(shell ruby -e 'require "rbconfig"; puts RbConfig::CONFIG["topdir"]') LOCAL_LDFLAGS=$(shell ruby -e 'require "rbconfig"; puts RbConfig::CONFIG["LIBRUBYARG"]') +LOCAL_LDFLAGS += -framework Ruby all: ESL.so Index: ruby/esl_wrap.cpp =================================================================== --- ruby/esl_wrap.cpp (revision 13432) +++ ruby/esl_wrap.cpp (working copy) @@ -823,7 +823,7 @@ -#include +#include /* Remove global macros defined in Ruby's win32.h */ #ifdef write Index: Makefile =================================================================== --- Makefile (revision 13432) +++ Makefile (working copy) @@ -12,7 +12,7 @@ OBJS=src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o SRC=src/esl.c src/esl_event.c src/esl_threadmutex.c src/esl_config.c src/esl_oop.cpp HEADERS=src/include/esl_config.h src/include/esl_event.h src/include/ esl.h src/include/esl_threadmutex.h src/include/esl_oop.h -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x # comment the next line to disable c++ (no swig mods for you then) OBJS += src/esl_oop.o /b On May 26, 2009, at 10:22 AM, dujinfang wrote: > Hi, > > Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library > On MacOSX 10.5, I can't get ESL for ruby work. make throws error: > > sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make > rubymod > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/ > Users/ > seven/workspace/test/freeswitch/trunk/libs/esl/src/include - > DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- > math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ > test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb - > I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C ruby > g++ -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall > -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c > esl_wrap.cpp -o esl_wrap.o > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. > Undefined symbols: > "_main", referenced from: > start in crt1.10.5.o > ld: symbol(s) not found > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > > Thanks for any help. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/182752a0/attachment.html From chris at maxpowersoft.com Tue May 26 08:38:54 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Tue, 26 May 2009 08:38:54 -0700 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> Message-ID: <4A1C0D0E.2030301@maxpowersoft.com> There is a cloud computing company named 3Tera (AppLogic) that does have an international presence and will keep your FreeSWITCH instance running on a dedicated server using Xen and HA. I spoke with one of their senior engineers about 1 month ago in regards to actually setting up an LCR scenario using their servers located in Europe and the United States. These guys are a little different in the cloud computing world and I believe closer fit the needs of a telephony application. As-is, there are companies using 3tera for their Asterisk installs. So if you want cloud computing with dedicated hardware resources and a set geographic location, then these guys do it. Kind of the best of both worlds. Just a quick 2 cents... Regards, Chris Brian West wrote: > > On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote: > >> Hey Brian, >> >> FreeSWITCH in EC2 is a bit of a mystery to me... >> >> Call me old fashioned but in my mind VoIP and geography are linked >> in %99 of scenarios. Having VoIP services in a pure "cloud" >> environment just doesn't sound like a good idea to me. >> >> Consider a "hosted" environment with clients registered to a >> FreeSWITCH server. One day your instance is physically running on >> hardware in Seattle. The next day it could (potentially) be running >> in Chicago. That's obviously a very different routing path for your >> clients. Even /if/ Amazon (or whomever) employs every routing/network >> trick in the book you still won't be able to get over that change in >> geography. > > For some people this isn't a huge difference... now if it were to swap > continents then yes it would be a problem. But I haven't seen Amazon > do this but I haven't left the instances up long enough to see. > >> It's certainly possible a change like this may very well never >> happen in practice. I wouldn't know; I've never used EC2 and I don't >> even know that much about it. I'm just curious how well strictly, >> practically speaking this will work in the long term. > > There are other companies that do this stuff but personally me... I > want my stuff running on real hardware. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/feecf661/attachment-0001.html From jcromes at gmail.com Tue May 26 08:46:21 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Tue, 26 May 2009 10:46:21 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> Message-ID: <4A1C0ECD.5090900@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/d53cc288/attachment.html From dujinfang at gmail.com Tue May 26 09:20:09 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 27 May 2009 00:20:09 +0800 Subject: [Freeswitch-users] how to enable ESL for ruby? In-Reply-To: <9D8ADB1E-123A-4347-8A31-B13BE3C760D3@freeswitch.org> References: <9D8ADB1E-123A-4347-8A31-B13BE3C760D3@freeswitch.org> Message-ID: Thanks Brain. Got ESL.so, however on my Mac it is #include instead of . But it can't find the ESL when I require 'ESL' in ruby. Even I put ESL.so in one of the dir of $: Any clue for me? seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl$ make rubymod cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes -c src/esl_event.c -o src/esl_event.o cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o src/ esl_threadmutex.o cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 - ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes -c src/esl_config.c -o src/esl_config.o g++ -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/ esl_config.o src/esl_oop.o ranlib libesl.a make MYLIB="../libesl.a" SOLINK="-dynamiclib -Xlinker -x" CFLAGS="-I/ Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/include - DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ test/freeswitch/oldtrunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C ruby g++ -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 -c esl_wrap.cpp -o esl_wrap.o g++ -dynamiclib -Xlinker -x esl_wrap.o ../libesl.a -lruby -framework Ruby -o ESL.so -L. seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl$ cd ruby/ seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl/ ruby$ ruby single_command.rb single_command.rb:3:in `require': no such file to load -- ESL (LoadError) from single_command.rb:3 seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl/ ruby$ irb irb(main):001:0> $: => ["/opt/local/lib/ruby/site_ruby/1.8", "/opt/local/lib/ruby/ site_ruby/1.8/i686-darwin9", "/opt/local/lib/ruby/site_ruby", "/opt/ local/lib/ruby/vendor_ruby/1.8", "/opt/local/lib/ruby/vendor_ruby/1.8/ i686-darwin9", "/opt/local/lib/ruby/vendor_ruby", "/opt/local/lib/ruby/ 1.8", "/opt/local/lib/ruby/1.8/i686-darwin9", "."] irb(main):002:0> On May 26, 2009, at 11:27 PM, Brian West wrote: > The makefile will have to be changed to work with OS X since the > linking is done differently. > > It would be very similar to this one http://www.bkw.org/esl.imac.diff > > Below will get it to compile: > > imac:esl brian$ svn diff > Index: ruby/Makefile > =================================================================== > --- ruby/Makefile (revision 13432) > +++ ruby/Makefile (working copy) > @@ -1,6 +1,7 @@ > ## no idea how to simply ask ruby which one to use > LOCAL_CFLAGS=-I$(shell ruby -e 'require "rbconfig"; puts > RbConfig::CONFIG["topdir"]') > LOCAL_LDFLAGS=$(shell ruby -e 'require "rbconfig"; puts > RbConfig::CONFIG["LIBRUBYARG"]') > +LOCAL_LDFLAGS += -framework Ruby > > all: ESL.so > > Index: ruby/esl_wrap.cpp > =================================================================== > --- ruby/esl_wrap.cpp (revision 13432) > +++ ruby/esl_wrap.cpp (working copy) > @@ -823,7 +823,7 @@ > > > > -#include > +#include > > /* Remove global macros defined in Ruby's win32.h */ > #ifdef write > Index: Makefile > =================================================================== > --- Makefile (revision 13432) > +++ Makefile (working copy) > @@ -12,7 +12,7 @@ > OBJS=src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o > SRC=src/esl.c src/esl_event.c src/esl_threadmutex.c src/ > esl_config.c src/esl_oop.cpp > HEADERS=src/include/esl_config.h src/include/esl_event.h src/ > include/esl.h src/include/esl_threadmutex.h src/include/esl_oop.h > -SOLINK=-shared -Xlinker -x > +SOLINK=-dynamiclib -Xlinker -x > # comment the next line to disable c++ (no swig mods for you then) > OBJS += src/esl_oop.o > > > > /b > > > > On May 26, 2009, at 10:22 AM, dujinfang wrote: > >> Hi, >> >> Following the wiki: http://wiki.freeswitch.org/wiki/Event_Socket_Library >> On MacOSX 10.5, I can't get ESL for ruby work. make throws error: >> >> sevens-mac-pro:~/workspace/test/freeswitch/trunk/libs/esl$ make >> rubymod >> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/ >> Users/ >> seven/workspace/test/freeswitch/trunk/libs/esl/src/include - >> DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- >> math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- >> prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ >> test/freeswitch/trunk/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb - >> I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" >> CXX_CFLAGS="" -C ruby >> g++ -I/Users/seven/workspace/test/freeswitch/trunk/libs/esl/src/ >> include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC - >> Wall >> -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686-darwin9 >> -c >> esl_wrap.cpp -o esl_wrap.o >> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. >> Undefined symbols: >> "_main", referenced from: >> start in crt1.10.5.o >> ld: symbol(s) not found >> collect2: ld returned 1 exit status >> make[1]: *** [ESL.so] Error 1 >> >> Thanks for any help. > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/2e7d50a9/attachment.html From erik at erikwickstrom.com Tue May 26 09:21:32 2009 From: erik at erikwickstrom.com (Erik Wickstrom) Date: Tue, 26 May 2009 09:21:32 -0700 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> Message-ID: <3d381e170905260921k68cbec02h27297c717c2fcd91@mail.gmail.com> I've been running a production FS app on EC2 since December. It's been really stable. Same server/instance since day1. We've haven't had any complaints.... Erik On Tue, May 26, 2009 at 8:11 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > On Tue, May 26, 2009 at 10:31 AM, Brian West wrote: > > Not with FreeSWITCH in our testing. Now if you have stupid defaults > > in your virtualization env. it might act funny but I have run FS on > > EC2 without a problem. > > > > /b > > Hey Brian, > > FreeSWITCH in EC2 is a bit of a mystery to me... > > Call me old fashioned but in my mind VoIP and geography are linked > in %99 of scenarios. Having VoIP services in a pure "cloud" > environment just doesn't sound like a good idea to me. > > Consider a "hosted" environment with clients registered to a > FreeSWITCH server. One day your instance is physically running on > hardware in Seattle. The next day it could (potentially) be running > in Chicago. That's obviously a very different routing path for your > clients. Even /if/ Amazon (or whomever) employs every routing/network > trick in the book you still won't be able to get over that change in > geography. > > It's certainly possible a change like this may very well never > happen in practice. I wouldn't know; I've never used EC2 and I don't > even know that much about it. I'm just curious how well strictly, > practically speaking this will work in the long term. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/4640d8f8/attachment-0001.html From brian at freeswitch.org Tue May 26 09:27:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 11:27:43 -0500 Subject: [Freeswitch-users] how to enable ESL for ruby? In-Reply-To: References: <9D8ADB1E-123A-4347-8A31-B13BE3C760D3@freeswitch.org> Message-ID: On May 26, 2009, at 11:20 AM, dujinfang wrote: > Thanks Brain. Got ESL.so, however on my Mac it is #include > instead of . Actually since we do -framework Ruby it should be ruby/ruby but I think the line above the -framework Ruby should be removed since you're doing i tthe Mac way. /b > > But it can't find the ESL when I require 'ESL' in ruby. Even I put > ESL.so in one of the dir of $: > > Any clue for me? > > seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl > $ make rubymod > cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o > cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes -c src/esl_event.c -o src/esl_event.o > cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o src/ > esl_threadmutex.o > cc -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes -c src/esl_config.c -o src/ > esl_config.o > g++ -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC - > Wall -Werror -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o > ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/ > esl_config.o src/esl_oop.o > ranlib libesl.a > make MYLIB="../libesl.a" SOLINK="-dynamiclib -Xlinker -x" CFLAGS="-I/ > Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/include - > DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast- > math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes" CXXFLAGS="-I/Users/seven/workspace/ > test/freeswitch/oldtrunk/libs/esl/src/include -DHAVE_EDITLINE -g - > ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused- > variable" CXX_CFLAGS="" -C ruby > g++ -I/Users/seven/workspace/test/freeswitch/oldtrunk/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC - > Wall -Werror -Wno-unused-variable -I/opt/local/lib/ruby/1.8/i686- > darwin9 -c esl_wrap.cpp -o esl_wrap.o > g++ -dynamiclib -Xlinker -x esl_wrap.o ../libesl.a -lruby -framework > Ruby -o ESL.so -L. > seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/esl > $ cd ruby/ > seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/ > esl/ruby$ ruby single_command.rb > single_command.rb:3:in `require': no such file to load -- ESL > (LoadError) > from single_command.rb:3 > seven at du-sevens-mac-pro:~/workspace/test/freeswitch/oldtrunk/libs/ > esl/ruby$ irb > irb(main):001:0> $: > => ["/opt/local/lib/ruby/site_ruby/1.8", "/opt/local/lib/ruby/ > site_ruby/1.8/i686-darwin9", "/opt/local/lib/ruby/site_ruby", "/opt/ > local/lib/ruby/vendor_ruby/1.8", "/opt/local/lib/ruby/vendor_ruby/ > 1.8/i686-darwin9", "/opt/local/lib/ruby/vendor_ruby", "/opt/local/ > lib/ruby/1.8", "/opt/local/lib/ruby/1.8/i686-darwin9", "."] > irb(main):002:0> Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/baf6dad7/attachment.html From damin at nacs.net Tue May 26 09:31:38 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Tue, 26 May 2009 12:31:38 -0400 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <4A1C0D0E.2030301@maxpowersoft.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> <4A1C0D0E.2030301@maxpowersoft.com> Message-ID: <006c01c9de1f$715ad360$54107a20$@net> I can say, from having met with and talked to the CEO and founder of Applogic that these guys are really revolutionary in their approach to cloud computing. I spoke on a panel w/ the founder at an ISPcon several years ago, and their approach is that of a utility company, treating computing resources like that of a power company. Cool stuff! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Danielson Sent: Tuesday, May 26, 2009 11:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in Amazon EC2 for production? There is a cloud computing company named 3Tera (AppLogic) that does have an international presence and will keep your FreeSWITCH instance running on a dedicated server using Xen and HA. I spoke with one of their senior engineers about 1 month ago in regards to actually setting up an LCR scenario using their servers located in Europe and the United States. These guys are a little different in the cloud computing world and I believe closer fit the needs of a telephony application. As-is, there are companies using 3tera for their Asterisk installs. So if you want cloud computing with dedicated hardware resources and a set geographic location, then these guys do it. Kind of the best of both worlds. Just a quick 2 cents... Regards, Chris Brian West wrote: On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote: Hey Brian, FreeSWITCH in EC2 is a bit of a mystery to me... Call me old fashioned but in my mind VoIP and geography are linked in %99 of scenarios. Having VoIP services in a pure "cloud" environment just doesn't sound like a good idea to me. Consider a "hosted" environment with clients registered to a FreeSWITCH server. One day your instance is physically running on hardware in Seattle. The next day it could (potentially) be running in Chicago. That's obviously a very different routing path for your clients. Even /if/ Amazon (or whomever) employs every routing/network trick in the book you still won't be able to get over that change in geography. For some people this isn't a huge difference... now if it were to swap continents then yes it would be a problem. But I haven't seen Amazon do this but I haven't left the instances up long enough to see. It's certainly possible a change like this may very well never happen in practice. I wouldn't know; I've never used EC2 and I don't even know that much about it. I'm just curious how well strictly, practically speaking this will work in the long term. There are other companies that do this stuff but personally me... I want my stuff running on real hardware. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _____ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- This message has been scanned for viruses and dangerous content by N2Net Mailshield, and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/e1ae129f/attachment.html From brian at freeswitch.org Tue May 26 09:36:23 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 11:36:23 -0500 Subject: [Freeswitch-users] Sounds order Message-ID: I'm getting ready for the next sound file order for FreeSWITCH. I have a rather large set of files to be recorded for the zRTP integration if anyone wants to help out. ;) Please contact me off list. I would like everyone to update and try out voicemail and nitpick anything that you feel is wrong there too and let me know so I can have them corrected in this order also. Thanks, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/a4966431/attachment-0001.html From lcm at marshap.com Tue May 26 09:49:47 2009 From: lcm at marshap.com (Larry Marshall) Date: Tue, 26 May 2009 09:49:47 -0700 Subject: [Freeswitch-users] Help understanding call_timeout in two dialplans Message-ID: <009f01c9de21$faffb980$f0ff2c80$@com> On inbound calls made to 5551212, which call_timeout will be active, 15 or 20? Is it the last hit? Is there a URL which describes all the applications, for example, export in the default.xml? Thanks for this amazing software, Lars In conf/dialplan/public/00_inbound.xml: In conf/dialplan/default.xml: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/641c0f00/attachment.html From brian at freeswitch.org Tue May 26 09:53:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 11:53:30 -0500 Subject: [Freeswitch-users] Help understanding call_timeout in two dialplans In-Reply-To: <009f01c9de21$faffb980$f0ff2c80$@com> References: <009f01c9de21$faffb980$f0ff2c80$@com> Message-ID: <09059257-6D14-48B1-8057-96C22DD3AD0C@freeswitch.org> last one set will win! /b On May 26, 2009, at 11:49 AM, Larry Marshall wrote: > On inbound calls made to 5551212, which call_timeout will be active, > 15 or 20? Is it the last hit? > > Is there a URL which describes all the applications, for example, > export in the default.xml? > > Thanks for this amazing software, Lars > > In conf/dialplan/public/00_inbound.xml: > > > expression="^1323(5551212)$"> > > > > > > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/30c651bd/attachment.html From fdelawarde at wirelessmundi.com Tue May 26 10:06:17 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 26 May 2009 19:06:17 +0200 Subject: [Freeswitch-users] Sounds order In-Reply-To: References: Message-ID: <1243357577.20111.31.camel@localhost.localdomain> Hi Brian, Is FreeSWITCH going to have Spanish/French/... sounds as well or do those need to be? Thanks, Fran?ois. On Tue, 2009-05-26 at 11:36 -0500, Brian West wrote: > I'm getting ready for the next sound file order for FreeSWITCH. I > have a rather large set of files to be recorded for the zRTP > integration if anyone wants to help out. ;) Please contact me off > list. I would like everyone to update and try out voicemail and > nitpick anything that you feel is wrong there too and let me know so I > can have them corrected in this order also. > > > > Thanks, > > > Brian West > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/49d2a564/attachment-0001.html From brian at freeswitch.org Tue May 26 10:11:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 12:11:39 -0500 Subject: [Freeswitch-users] Sounds order In-Reply-To: <1243357577.20111.31.camel@localhost.localdomain> References: <1243357577.20111.31.camel@localhost.localdomain> Message-ID: At this time we only have english it will take $1200-$2000 to record each language. /b On May 26, 2009, at 12:06 PM, Fran?ois Delawarde wrote: > Hi Brian, > > Is FreeSWITCH going to have Spanish/French/... sounds as well or do > those need to be? > > Thanks, > Fran?ois. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/0c8f4ff6/attachment.html From nas at nas.lv Tue May 26 10:19:46 2009 From: nas at nas.lv (Andrey Nepomnyaschih) Date: Tue, 26 May 2009 13:19:46 -0400 Subject: [Freeswitch-users] NOTIFY Messages Message-ID: <8B55809D6485CD448855DFDDC47D00071D2DD620A7@VMBX103.ihostexchange.net> Hello everyone, Does anyone know is it possible to disable NOTIFY messages coming out of FreeSwitch for particular endpoint? The reason I'm asking it is because I have a gateway (D-Link DVG-7111S) that reboots when it receives such a packet. Kind Regards, Andrey Nepomnyaschih -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/3220146c/attachment.html From brian at freeswitch.org Tue May 26 10:28:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 12:28:17 -0500 Subject: [Freeswitch-users] NOTIFY Messages In-Reply-To: <8B55809D6485CD448855DFDDC47D00071D2DD620A7@VMBX103.ihostexchange.net> References: <8B55809D6485CD448855DFDDC47D00071D2DD620A7@VMBX103.ihostexchange.net> Message-ID: You can set this param on a profile This should take care of it but really that gateway is broken if it reboots on a MWI notify. /b On May 26, 2009, at 12:19 PM, Andrey Nepomnyaschih wrote: > Hello everyone, > > Does anyone know is it possible to disable NOTIFY messages coming > out of FreeSwitch for particular endpoint? The reason I?m asking it > is because I have a gateway (D-Link DVG-7111S) that reboots when it > receives such a packet. > > Kind Regards, > Andrey Nepomnyaschih > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/389d798c/attachment.html From brian at freeswitch.org Tue May 26 10:34:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 12:34:48 -0500 Subject: [Freeswitch-users] STUN and rport on Polycom phones Message-ID: <74FB7C03-EC28-4935-9C13-F25652704343@freeswitch.org> It has come to my attention that Polycom hasn't had a business case to support rport and STUN. If you can please kindly email Marek.Dutkiewicz at polycom.com and let him know you would like to see STUN and rport support in the polycom products. Its really one of the last things missing in the phone to make it easy to deploy in a NAT env. Thanks, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/5e965a87/attachment.html From nas at nas.lv Tue May 26 11:04:00 2009 From: nas at nas.lv (Andrey Nepomnyaschih) Date: Tue, 26 May 2009 14:04:00 -0400 Subject: [Freeswitch-users] NOTIFY Messages In-Reply-To: References: <8B55809D6485CD448855DFDDC47D00071D2DD620A7@VMBX103.ihostexchange.net> Message-ID: <8B55809D6485CD448855DFDDC47D00071D2DD620B3@VMBX103.ihostexchange.net> Well, that didn't help. I've tried to put it into endpoint (conf/directory/default/*.xml) configuration or profile (conf/sip_profiles/internal.xml) without a success. Is there any other option? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, May 26, 2009 9:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] NOTIFY Messages You can set this param on a profile This should take care of it but really that gateway is broken if it reboots on a MWI notify. /b On May 26, 2009, at 12:19 PM, Andrey Nepomnyaschih wrote: Hello everyone, Does anyone know is it possible to disable NOTIFY messages coming out of FreeSwitch for particular endpoint? The reason I'm asking it is because I have a gateway (D-Link DVG-7111S) that reboots when it receives such a packet. Kind Regards, Andrey Nepomnyaschih _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/b88231e1/attachment-0001.html From msc at freeswitch.org Tue May 26 11:26:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 May 2009 11:26:05 -0700 Subject: [Freeswitch-users] How to work with the debug logfile? In-Reply-To: <5e414ed0905250819o792c42f9j66fe3c70730a89d0@mail.gmail.com> References: <5e414ed0905250819o792c42f9j66fe3c70730a89d0@mail.gmail.com> Message-ID: <87f2f3b90905261126r74047fc6n3f12e1c2a49205a2@mail.gmail.com> On Mon, May 25, 2009 at 8:19 AM, Dennis wrote: > hi, > > we encounter some small problems withing the past 2 days and we are > trying to find out more about the problems. for this we downloaded the > debug logfiles written by fs, but we do not manage to filter all > log-entries for one single special call. > > we are using socket outbound and would like to see all entries > (inbound/outbound) of one call. > > is this possible? > > kind regards > dennis > This isn't the easiest thing to do but it is possible. In the past I've used a combination of uuid and channel name with grep. Also, I recommend rotating log files using "fsctl send_sighup" which helps keep you from getting a 10GB freeswitch.log file. :) The first thing I do is find the call's uuid, either by manually searching through the log file or by using a CDR. (Use whichever works best for you.) Then use grep to find all the line numbers for that particular uuid. Here's a sample from my Mac where I used portaudio to call the main conference line: grep -n b1f9e5ce-29a6-497a-a628-71f1bdefa0cb freeswitch.log In this case I found out my channel name is "portaudio/ sip:888 at conference.freeswitch.org " so I grep for it: grep "portaudio/sip:888 at conference.freeswitch.org" freeswitch.log It's not perfect but it helps narrow down your call. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/7ce8e15f/attachment.html From msc at freeswitch.org Tue May 26 12:02:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 May 2009 12:02:01 -0700 Subject: [Freeswitch-users] IVR xml dialplan for outbound calls In-Reply-To: References: Message-ID: <87f2f3b90905261202u379045a6p12091ec9fd7899d5@mail.gmail.com> This is totally possible. You need to look at the originate command. Also, if you have an IVR or a dialplan extension that does what you want - plays prompts, accepts DTMF digits from caller, etc. - then you can use it with inbound or outbound calls. For an outbound call just route it to the dp extension. For example to route an outbound call to the demo IVR do this: originate sofia/gateway/mygatewayname/18005551212 5000 Have fun. -MC On Mon, May 25, 2009 at 10:03 PM, Paul Li wrote: > There are a few examples in the wiki, showing how to configure IVR for > inbound calls. My question lies in whether it is possible to write a > dialplan in xml or scripts to configure IVR for outbound calls. Here is a > typical scenario (of usage): > > (1) Call any extension or external endpoints (maybe PSTN numbers); > (2) Activate IVR menu once the recipient answers call; > (3) Take feedback (DTMF keystrokes or survey recordings). > > Your assistance is highly appreciated! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/5b1d38c9/attachment.html From msc at freeswitch.org Tue May 26 12:06:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 May 2009 12:06:54 -0700 Subject: [Freeswitch-users] Sounds order In-Reply-To: <1243357577.20111.31.camel@localhost.localdomain> References: <1243357577.20111.31.camel@localhost.localdomain> Message-ID: <87f2f3b90905261206y22b3924bj736e68b018fde22c@mail.gmail.com> On Tue, May 26, 2009 at 10:06 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Hi Brian, > > Is FreeSWITCH going to have Spanish/French/... sounds as well or do those > need to be? > We have a volunteer who is going to record a set of Spanish prompts. These would probably be classified as Spanish Latin-America as opposed to Castillian. We could use a volunteer for French prompts. Or if there is a very generous benefactor who is willing to donate we could have GM Voices do high-quality French prompts. -MC > > > Thanks, > Fran?ois. > > On Tue, 2009-05-26 at 11:36 -0500, Brian West wrote: > > I'm getting ready for the next sound file order for FreeSWITCH. I have a > rather large set of files to be recorded for the zRTP integration if anyone > wants to help out. ;) Please contact me off list. I would like everyone > to update and try out voicemail and nitpick anything that you feel is wrong > there too and let me know so I can have them corrected in this order also. > > > > Thanks, > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/e1265bd6/attachment.html From brian at freeswitch.org Tue May 26 12:47:52 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 14:47:52 -0500 Subject: [Freeswitch-users] Pre8 Release on Digg Message-ID: Dear FreeSWITCHers, Now I'm gonna take a moment here to guilt each and everyone of you into checking out the story about Pre8 on Digg. We have all worked long and hard to get to 1.0.4 and we still have a little bit to go. So everyone out there that asks "What can I do to help the project?", this is your chance to do so. Help us to promote the project, which brings more people to help in supporting the community, the project and you the end user. Also looking for people to help manage jira, test bugs, ask the right questions and line up the bugs so we can knock them out! Please email me if you're interested. Here is the link for you to help out http://digg.com/search?s=FreeSWITCH+Pre8 Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/864e78b1/attachment.html From diego.viola at gmail.com Tue May 26 13:03:37 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 26 May 2009 16:03:37 -0400 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: Message-ID: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> Digged. On Tue, May 26, 2009 at 3:47 PM, Brian West wrote: > Dear FreeSWITCHers, > Now I'm gonna take a moment here to guilt each and everyone of you into > checking out the story about Pre8 on Digg. ?We have all worked long and hard > to get to 1.0.4 and we still have a little bit to go. ?So everyone out there > that asks "What can I do to help the project?", this is your chance to do > so. ?Help us to promote the project, which brings more people to help in > supporting the community, the project and you the end user. > Also looking for people to help manage jira, test bugs, ask the right > questions and line up the bugs so we can knock them out! Please email me if > you're interested. > Here is the link for you to help > out?http://digg.com/search?s=FreeSWITCH+Pre8 > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From drew.ozier at gmail.com Tue May 26 13:36:40 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Tue, 26 May 2009 16:36:40 -0400 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH Message-ID: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> I've got a configuration where I receive inbound calls and dial out to a pre-determined 800-number based on the DNIS of the call. I set and have everything set up so that DTMF only comes to me via inband. When I'm providing DTMF data to the IVR, it will recognize a single keypress as a double-tap. My FreeSWITCH logs only contain one DTMF entry, but when I listen to the receiving end of the call, I can hear a hiccup in the DTMF tone that is getting played. When I do not use 'start_dtmf', this problem goes away. I need inband DTMF detection, but I can't have it messing up the audio stream. Any thoughts? -Drew Ozier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/0f8016e1/attachment.html From grevenx at me.com Tue May 26 13:40:38 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Tue, 26 May 2009 22:40:38 +0200 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> References: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> Message-ID: <9E7EA58F-F946-445D-AAC9-EFEF21185D71@me.com> Diggedy dug! On 26. mai. 2009, at 22.03, Diego Viola wrote: > Digged. > > On Tue, May 26, 2009 at 3:47 PM, Brian West > wrote: >> Dear FreeSWITCHers, >> Now I'm gonna take a moment here to guilt each and everyone of you >> into >> checking out the story about Pre8 on Digg. We have all worked long >> and hard >> to get to 1.0.4 and we still have a little bit to go. So everyone >> out there >> that asks "What can I do to help the project?", this is your chance >> to do >> so. Help us to promote the project, which brings more people to >> help in >> supporting the community, the project and you the end user. >> Also looking for people to help manage jira, test bugs, ask the right >> questions and line up the bugs so we can knock them out! Please >> email me if >> you're interested. >> Here is the link for you to help >> out http://digg.com/search?s=FreeSWITCH+Pre8 >> Brian West >> brian at freeswitch.org >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue May 26 13:42:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 15:42:56 -0500 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: <9E7EA58F-F946-445D-AAC9-EFEF21185D71@me.com> References: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> <9E7EA58F-F946-445D-AAC9-EFEF21185D71@me.com> Message-ID: Thank you... Now please tell 10 of your friends about FreeSWITCH ;) /b On May 26, 2009, at 3:40 PM, Even Andr? Fiskvik wrote: > Diggedy dug! > > On 26. mai. 2009, at 22.03, Diego Viola wrote: Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/3dccbad6/attachment.html From grevenx at me.com Tue May 26 14:11:27 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Tue, 26 May 2009 23:11:27 +0200 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> <9E7EA58F-F946-445D-AAC9-EFEF21185D71@me.com> Message-ID: ...or be followed with 10 yrs of bad luck and hardware failures! On 26. mai. 2009, at 22.42, Brian West wrote: > Thank you... Now please tell 10 of your friends about FreeSWITCH ;) > > /b > > On May 26, 2009, at 3:40 PM, Even Andr? Fiskvik wrote: > >> Diggedy dug! >> >> On 26. mai. 2009, at 22.03, Diego Viola wrote: > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/265cacd6/attachment.html From brian at freeswitch.org Tue May 26 14:16:57 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 16:16:57 -0500 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> <9E7EA58F-F946-445D-AAC9-EFEF21185D71@me.com> Message-ID: <7583BD1A-A66B-40A1-BA5A-E3FB48CBBA99@freeswitch.org> Remember Save the cheerlead, save the world? In this case Digg the story, get your bugs fixed? /b On May 26, 2009, at 4:11 PM, Even Andr? Fiskvik wrote: > ...or be followed with 10 yrs of bad luck and hardware failures! > > On 26. mai. 2009, at 22.42, Brian West wrote: Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/d90448d5/attachment.html From codecomplete at free.fr Tue May 26 14:24:05 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 26 May 2009 14:24:05 -0700 (PDT) Subject: [Freeswitch-users] PBX on a (Sheeva|Pogo)Plug? Message-ID: <23731619.post@talk.nabble.com> Hello I was wondering if someone had successfully ran a PBX on those tiny, $100 devices that run Linux? www.pogoplug.com www.plugcomputer.org I'm thinking of hooking up a Sangoma U100 (http://wiki.sangoma.com/sangoma-wanpipe-usbfxo) to the USB port so it can handle a couple of analog lines. And installing a compact LAMP server so the user can manage the PBX server from a web browser. What do you think? Thank you. -- View this message in context: http://www.nabble.com/PBX-on-a-%28Sheeva%7CPogo%29Plug--tp23731619p23731619.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dave at 3c.co.uk Tue May 26 15:57:07 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 26 May 2009 23:57:07 +0100 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted through FreeSWITCH References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> Message-ID: <7753D7DF98684725AE7BA4A638564F4A@DELL9> Hi Drew, When you say that the problem goes away if you don't use start_dtmf, do you mean that you get one tone recognised per tone or none? If the former, then you've got DTMF being signalled out of band as well; in that case, why do you need inband detection? --Dave ----- Original Message ----- From: Drew Ozier To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, May 26, 2009 9:36 PM Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted through FreeSWITCH I've got a configuration where I receive inbound calls and dial out to a pre-determined 800-number based on the DNIS of the call. I set and have everything set up so that DTMF only comes to me via inband. When I'm providing DTMF data to the IVR, it will recognize a single keypress as a double-tap. My FreeSWITCH logs only contain one DTMF entry, but when I listen to the receiving end of the call, I can hear a hiccup in the DTMF tone that is getting played. When I do not use 'start_dtmf', this problem goes away. I need inband DTMF detection, but I can't have it messing up the audio stream. Any thoughts? -Drew Ozier ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/38bd7b1f/attachment.html From anthony.minessale at gmail.com Tue May 26 16:56:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 May 2009 18:56:44 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A1C0ECD.5090900@gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> Message-ID: <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> the easiest way would be the new feature I added to 13442 in the conference profile add to your and in your dialplan or Don't forget the wishlist and donate button on the main site.... On Tue, May 26, 2009 at 10:46 AM, wrote: > Ok, that sounds doable... I have no problem banging around with some > code. Thanks for the advice... > I'm new to FIFOs and FreeSwitch in general, so please check my logic > here... > > - When a user enters their conference number, I check to see if that > conference exists (because the conference shouldn't exist until the leader > enters). > - If that conference exists, the user gets dropped straight into it. > - If that conference does not exist, I park that user into a FIFO, with > music. > > - When the leader enters their conference number and password, I put them > into that conference (which creates the conference). > - Then check for the presence of a FIFO with parked users. > - If the FIFO contains parked users, I transfer everyone from the FIFO into > the conference. > > Is that what you were thinking? > > > > Brian West wrote: > > Jason, > There are many ways to accomplish this using FreeSWITCH. All of > which will require you to do a little bit of coding in js, lua or some > other language. > > 1. Park all callers into a fifo.. (see mod_fifo) > 2. When leader auths in your script then you uuid_transfer them all > into the conference. > > You could also do this via esl using perl, lua, python, php or ruby. > > That should accomplish the same ending result. > > /b > > > > On May 26, 2009, at 9:38 AM, Jason Cromes wrote: > > > > This is easy in Asterisk because you can pop someone into MeetMe with > different flags. So, in my IVR, I prompt for the "conference number" > (known to all) and then the "password" (known only to the owner/ > leader). > If the proper password is entered, that user is sent to conference XYZ > with the leader flag set. If no password is entered, the user goes to > conference XYZ, without the leader flag. If anyone enters before the > leader, they're told by MeetMe that "the conference will begin when > the > leader arrives" and MeetMe provides MOH until that time. > > Help! This is an absolute deal-breaker for my install... How can I > do > this in FreeSwitch? > Thanks... > > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/fbe82533/attachment-0001.html From larclap at yahoo.com Tue May 26 17:03:15 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 26 May 2009 17:03:15 -0700 Subject: [Freeswitch-users] Questions on build 13441 Message-ID: <000301c9de5e$88c51800$9a4f4800$@com> I followed Michael Collin's instruction in an earlier email about building pre8. During the ./bootstrap.h, I encountered the following error: Entering directory /usr/src/freeswitch.trunk/libs/libsndfile Creating aclocal.m4 Running libtoolize... Putting files in AC_CONFIG_AUX_DIR, `Cfg'. Creating configure configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION Do I need to worry about this? Also, in general, do the XML files in config get updated during the 'make install', or are they left as they were from the previous builds? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/6f6f09f8/attachment.html From brian at freeswitch.org Tue May 26 17:17:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 26 May 2009 19:17:34 -0500 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <000301c9de5e$88c51800$9a4f4800$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> Message-ID: <20C7C516-7029-4F3A-A5CA-0F85BEC9205B@freeswitch.org> First off pre8 came prebootstrapped no need to run it. So please try again. /b On May 26, 2009, at 7:03 PM, Lars Zeb wrote: > I followed Michael Collin?s instruction in an earlier email about > building pre8. During the ./bootstrap.h, I encountered the following > error: > > Entering directory /usr/src/freeswitch.trunk/libs/libsndfile > Creating aclocal.m4 > Running libtoolize... > Putting files in AC_CONFIG_AUX_DIR, `Cfg'. > Creating configure > configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD > If this token and others are legitimate, please use > m4_pattern_allow. > See the Autoconf documentation. > configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION > > Do I need to worry about this? > > Also, in general, do the XML files in config get updated during the > ?make install?, or are they left as they were from the previous > builds? > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/91e63c66/attachment.html From msc at freeswitch.org Tue May 26 17:55:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 May 2009 17:55:15 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> Message-ID: <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > the easiest way would be the new feature I added to 13442 > > in the conference profile add > > > > to your > > and in your dialplan > > > > > or > > > > > Don't forget the wishlist and donate button on the main site.... > > And the wiki tax if you feel comfortable adding this to the wiki. If not let me know and I'll add it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/2c070a59/attachment.html From msc at freeswitch.org Tue May 26 17:57:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 26 May 2009 17:57:17 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <000301c9de5e$88c51800$9a4f4800$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> Message-ID: <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> > Also, in general, do the XML files in config get updated during the ?make > install?, or are they left as they were from the previous builds? > Running "make install" or "make samples" will not overwrite your existing configuration files. NOTE: This means that when the default configuration changes you will need to update your files manually and integrate the new changes. This is one reason why I recommend making as few edits as possible to the default configuration files. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/59f87f0f/attachment.html From diego.viola at gmail.com Tue May 26 18:06:53 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 26 May 2009 21:06:53 -0400 Subject: [Freeswitch-users] mod_nibblebill question In-Reply-To: <86a32abc0905101206j76c10a21oe059b360118e5254@mail.gmail.com> References: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> <86a32abc0905081208u76345237q190559b8f0b161d4@mail.gmail.com> <86a32abc0905101206j76c10a21oe059b360118e5254@mail.gmail.com> Message-ID: <86a32abc0905261806n556455cbie63eaa71786ac3cf@mail.gmail.com> Hi, Darren just added this today, in case if someone is interested. http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_cash_is_depleted Thanks Darren :). Regards, Diego On Sun, May 10, 2009 at 3:06 PM, Diego Viola wrote: > Darren Schreiber to me > That won't work. The code isn't written yet. Give mea week or two to > finish that. > > > On Fri, May 8, 2009 at 3:08 PM, Diego Viola wrote: >> I have set these actions: >> >> >> >> >> But when it reaches 0 cash it doesn't hangup :(. >> >> On Fri, May 8, 2009 at 2:34 PM, Diego Viola wrote: >>> Oh I see that it has it already :D >>> >>> >>> >>> >>> - >>> >>> >>> >>> >>> Diego >>> >>> On Fri, May 8, 2009 at 1:20 PM, Diego Viola wrote: >>>> Hi everyone, >>>> >>>> I'm currently developing a calling card application that uses event >>>> socket and mod_nibblebill to bill calls. Well, the question is: can >>>> mod_nibblebill disconnect a call when the balance is depleted, or when >>>> it reaches 0 cash? >>>> >>>> The wiki says: >>>> >>>> "Allow for disconnecting or re-routing calls when balance is depleted." >>>> >>>> But then at the bottom in "future goals" it says: >>>> >>>> "We don't yet warn or terminate calls when they've reached their limits." >>>> >>>> So I'm confused, I also don't see any options in how to set that. In >>>> case that mod_nibblebill doesn't support this yet, how hard would it >>>> be to add this? I'm willing to do it, I guess it's a variable on the >>>> XML config and then read that from the mod_nibblebill.c, and do some >>>> logic there. >>>> >>>> Just wondering if anyone had some experience with this, and if the >>>> feature is already there. >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>> >> > From jason at jasonjgw.net Tue May 26 18:15:10 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 27 May 2009 11:15:10 +1000 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: <86a32abc0905261303v51e005fg72a85b11ac58ad78@mail.gmail.com> <9E7EA58F-F946-445D-AAC9-EFEF21185D71@me.com> Message-ID: <20090527011510.GA16093@jdc.jasonjgw.net> Brian West wrote: > Thank you... Now please tell 10 of your friends about FreeSWITCH ;) Also, if you're a member of a Linux user's group or similar organization, now might be a good opportunity to raise FreeSWITCH awareness on their mailing list or at a meeting. From stevecrozz at gmail.com Tue May 26 18:17:26 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Tue, 26 May 2009 18:17:26 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> Message-ID: <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> Michael, Pardon me for hopping on this thread, but can you explain more about this new feature? I've been wanting something like this to apply different behaviors for different conference members. Can this be used to provide a 'moderator' with different behaviors bound to DTMF keys than regular callers? --Stephen On Tue, May 26, 2009 at 5:55 PM, Michael Collins wrote: > > > On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> the easiest way would be the new feature I added to 13442 >> >> in the conference profile add >> >> >> >> to your >> >> and in your dialplan >> >> >> >> >> or >> >> >> >> >> Don't forget the wishlist and donate button on the main site.... >> >> > And the wiki tax if you feel comfortable adding this to the wiki. If not > let me know and I'll add it. > -MC > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/a66d6b3a/attachment-0001.html From diego.viola at gmail.com Tue May 26 18:33:54 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 26 May 2009 21:33:54 -0400 Subject: [Freeswitch-users] XML config error Message-ID: <86a32abc0905261833l160cbc8fy96a3b1ee68f45a96@mail.gmail.com> Hi, I have downloaded the latest freeswitch trunk, and when I do reloadxml I get this. Error [unterminated ${var}] in line /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml line 12 Any ideas? I haven't edited that file myself. Thanks, Diego From larclap at yahoo.com Tue May 26 18:50:43 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 26 May 2009 18:50:43 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <20C7C516-7029-4F3A-A5CA-0F85BEC9205B@freeswitch.org> References: <000301c9de5e$88c51800$9a4f4800$@com> <20C7C516-7029-4F3A-A5CA-0F85BEC9205B@freeswitch.org> Message-ID: <002f01c9de6d$8c2e0ec0$a48a2c40$@com> So your saying instead of: cd /usr/src/freeswitch.trunk make clean svn up ../bootstrap.sh ../configure make install do: cd /usr/src/freeswitch.trunk make clean svn up ../configure make install ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, May 26, 2009 5:18 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions on build 13441 First off pre8 came prebootstrapped no need to run it. So please try again. /b On May 26, 2009, at 7:03 PM, Lars Zeb wrote: I followed Michael Collin's instruction in an earlier email about building pre8. During the ./bootstrap.h, I encountered the following error: Entering directory /usr/src/freeswitch.trunk/libs/libsndfile Creating aclocal.m4 Running libtoolize... Putting files in AC_CONFIG_AUX_DIR, `Cfg'. Creating configure configure.ac:278: error: possibly undefined macro: AC_OCTAVE_BUILD If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. configure.ac:478: error: possibly undefined macro: AC_GCC_VERSION Do I need to worry about this? Also, in general, do the XML files in config get updated during the 'make install', or are they left as they were from the previous builds? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090526/830ed584/attachment.html From jason at jasonjgw.net Tue May 26 19:16:35 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 27 May 2009 12:16:35 +1000 Subject: [Freeswitch-users] XML config error In-Reply-To: <86a32abc0905261833l160cbc8fy96a3b1ee68f45a96@mail.gmail.com> References: <86a32abc0905261833l160cbc8fy96a3b1ee68f45a96@mail.gmail.com> Message-ID: <20090527021635.GA19575@jdc.jasonjgw.net> Diego Viola wrote: > Hi, I have downloaded the latest freeswitch trunk, and when I do > reloadxml I get this. > > Error [unterminated ${var}] in line > /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml > line 12 > > Any ideas? I haven't edited that file myself. Have a look in that file to see whether there's a syntax error there. If so, and if it's in the source tree, fix it and submit a patch. From jcromes at gmail.com Tue May 26 19:19:13 2009 From: jcromes at gmail.com (j3flight) Date: Tue, 26 May 2009 19:19:13 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> Message-ID: <23734920.post@talk.nabble.com> Anthony - unbelievable! Thank you so much for implementing that! I kept going through the possibilities of using a FIFO, putting the javascript in a polling loop, or having everyone enter the conference muted and manually playing MOH. This feature absolutely makes my code a snap... Yes, I will remember the wish-list and I will update the wiki (probably tomorrow though). Stephen: This new feature doesn't provide the ability you're referring to... It gives me the ability to make normal users "wait" on a moderator to open the conference. However, the capability you're asking for already exists in FreeSwitch by choosing different "profiles" for your conferences. Within the mod_conference config file you can create different profiles which use different "caller controls" a.k.a. "in-conference dtmf features". So, create one profile for your normal users and another identical profile (except for the caller controls) for your moderators/admins. When you put them into the conference, you can select your profile like so: I hope that's all correct - again, I'm new. Anthony, I will try this out first thing tomorrow! -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23734920.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From diego.viola at gmail.com Tue May 26 19:24:00 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 26 May 2009 22:24:00 -0400 Subject: [Freeswitch-users] XML config error In-Reply-To: <20090527021635.GA19575@jdc.jasonjgw.net> References: <86a32abc0905261833l160cbc8fy96a3b1ee68f45a96@mail.gmail.com> <20090527021635.GA19575@jdc.jasonjgw.net> Message-ID: <86a32abc0905261924y7a274f04id62d639df908fcc4@mail.gmail.com> I just updated it, it was a bug that got fixed already. 22:19 <@bkw__> diegoviola: already fixed 22:19 <@bkw__> update 22:19 <@bkw__> close the } 22:20 <@bkw__> it was a bug we fixed already this morning that catches unclosed global preprocess vars Thanks, Diego On Tue, May 26, 2009 at 10:16 PM, Jason White wrote: > Diego Viola wrote: >> Hi, I have downloaded the latest freeswitch trunk, and when I do >> reloadxml I get this. >> >> Error [unterminated ${var}] in line >> /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml >> line 12 >> >> Any ideas? I haven't edited that file myself. > > Have a look in that file to see whether there's a syntax error there. If so, > and if it's in the source tree, fix it and submit a patch. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Tue May 26 19:51:40 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 26 May 2009 22:51:40 -0400 Subject: [Freeswitch-users] mod_nibblebill question In-Reply-To: <86a32abc0905261806n556455cbie63eaa71786ac3cf@mail.gmail.com> References: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> <86a32abc0905081208u76345237q190559b8f0b161d4@mail.gmail.com> <86a32abc0905101206j76c10a21oe059b360118e5254@mail.gmail.com> <86a32abc0905261806n556455cbie63eaa71786ac3cf@mail.gmail.com> Message-ID: <86a32abc0905261951j68c2a784maf2a1e89c0b77d61@mail.gmail.com> Sorry, this is the link. http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_the_balance_is_depleted Diego On Tue, May 26, 2009 at 9:06 PM, Diego Viola wrote: > Hi, > > Darren just added this today, in case if someone is interested. > > http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_cash_is_depleted > > Thanks Darren :). > > Regards, > > Diego > > On Sun, May 10, 2009 at 3:06 PM, Diego Viola wrote: >> Darren Schreiber to me >> That won't work. The code isn't written yet. Give mea week or two to >> finish that. >> >> >> On Fri, May 8, 2009 at 3:08 PM, Diego Viola wrote: >>> I have set these actions: >>> >>> >>> >>> >>> But when it reaches 0 cash it doesn't hangup :(. >>> >>> On Fri, May 8, 2009 at 2:34 PM, Diego Viola wrote: >>>> Oh I see that it has it already :D >>>> >>>> >>>> >>>> >>>> - >>>> >>>> >>>> >>>> >>>> Diego >>>> >>>> On Fri, May 8, 2009 at 1:20 PM, Diego Viola wrote: >>>>> Hi everyone, >>>>> >>>>> I'm currently developing a calling card application that uses event >>>>> socket and mod_nibblebill to bill calls. Well, the question is: can >>>>> mod_nibblebill disconnect a call when the balance is depleted, or when >>>>> it reaches 0 cash? >>>>> >>>>> The wiki says: >>>>> >>>>> "Allow for disconnecting or re-routing calls when balance is depleted." >>>>> >>>>> But then at the bottom in "future goals" it says: >>>>> >>>>> "We don't yet warn or terminate calls when they've reached their limits." >>>>> >>>>> So I'm confused, I also don't see any options in how to set that. In >>>>> case that mod_nibblebill doesn't support this yet, how hard would it >>>>> be to add this? I'm willing to do it, I guess it's a variable on the >>>>> XML config and then read that from the mod_nibblebill.c, and do some >>>>> logic there. >>>>> >>>>> Just wondering if anyone had some experience with this, and if the >>>>> feature is already there. >>>>> >>>>> Thanks, >>>>> >>>>> Diego >>>>> >>>> >>> >> > From ryder86 at googlemail.com Tue May 26 23:16:10 2009 From: ryder86 at googlemail.com (=?KOI8-R?B?4dLUxc0g98HTyczYxdc=?=) Date: Wed, 27 May 2009 10:16:10 +0400 Subject: [Freeswitch-users] Unicast isn't working Message-ID: Of course, it will be useful to try it. Artem > Hi Artem, > > Please to see that some of the stuff I wrote is useful to someone..! > > I've written an FS module which will send the audio over - it's more > efficient than using unicast. Let me know if you'd like a copy. > > Cheers -- > > Dave > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/0cd69beb/attachment-0001.html From grevenx at me.com Tue May 26 23:49:22 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Wed, 27 May 2009 08:49:22 +0200 Subject: [Freeswitch-users] mod_nibblebill question In-Reply-To: <86a32abc0905261951j68c2a784maf2a1e89c0b77d61@mail.gmail.com> References: <86a32abc0905081020j69104ec2of7fb3db8f59153a3@mail.gmail.com> <86a32abc0905081134l7f192317t37b56ee4c387ead6@mail.gmail.com> <86a32abc0905081208u76345237q190559b8f0b161d4@mail.gmail.com> <86a32abc0905101206j76c10a21oe059b360118e5254@mail.gmail.com> <86a32abc0905261806n556455cbie63eaa71786ac3cf@mail.gmail.com> <86a32abc0905261951j68c2a784maf2a1e89c0b77d61@mail.gmail.com> Message-ID: <9BB95A21-8CE7-417A-B608-91B5225293F2@me.com> Cool, looking forward to test this :) Even Andr? On 27. mai. 2009, at 04.51, Diego Viola wrote: > Sorry, this is the link. > > http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_the_balance_is_depleted > > Diego > > On Tue, May 26, 2009 at 9:06 PM, Diego Viola > wrote: >> Hi, >> >> Darren just added this today, in case if someone is interested. >> >> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_cash_is_depleted >> >> Thanks Darren :). >> >> Regards, >> >> Diego >> >> On Sun, May 10, 2009 at 3:06 PM, Diego Viola >> wrote: >>> Darren Schreiber to me >>> That won't work. The code isn't written yet. Give mea week or two to >>> finish that. >>> >>> >>> On Fri, May 8, 2009 at 3:08 PM, Diego Viola >>> wrote: >>>> I have set these actions: >>>> >>>> >>>> >>>> >>>> But when it reaches 0 cash it doesn't hangup :(. >>>> >>>> On Fri, May 8, 2009 at 2:34 PM, Diego Viola >>>> wrote: >>>>> Oh I see that it has it already :D >>>>> >>>>> >>>>> >>>>> >>>>> - >>>>> >>>>> >>>>> >>>>> >>>>> Diego >>>>> >>>>> On Fri, May 8, 2009 at 1:20 PM, Diego Viola >>>>> wrote: >>>>>> Hi everyone, >>>>>> >>>>>> I'm currently developing a calling card application that uses >>>>>> event >>>>>> socket and mod_nibblebill to bill calls. Well, the question is: >>>>>> can >>>>>> mod_nibblebill disconnect a call when the balance is depleted, >>>>>> or when >>>>>> it reaches 0 cash? >>>>>> >>>>>> The wiki says: >>>>>> >>>>>> "Allow for disconnecting or re-routing calls when balance is >>>>>> depleted." >>>>>> >>>>>> But then at the bottom in "future goals" it says: >>>>>> >>>>>> "We don't yet warn or terminate calls when they've reached >>>>>> their limits." >>>>>> >>>>>> So I'm confused, I also don't see any options in how to set >>>>>> that. In >>>>>> case that mod_nibblebill doesn't support this yet, how hard >>>>>> would it >>>>>> be to add this? I'm willing to do it, I guess it's a variable >>>>>> on the >>>>>> XML config and then read that from the mod_nibblebill.c, and do >>>>>> some >>>>>> logic there. >>>>>> >>>>>> Just wondering if anyone had some experience with this, and if >>>>>> the >>>>>> feature is already there. >>>>>> >>>>>> Thanks, >>>>>> >>>>>> Diego >>>>>> >>>>> >>>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Wed May 27 02:46:26 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 27 May 2009 11:46:26 +0200 Subject: [Freeswitch-users] Origionate a call via event_socket. relation between job_uuid and uuid Message-ID: <4A1D0BF2.4000800@gmx.net> I want to do the following: Originate a call via event_socket, I get back a job_uuid. Then I want to control the call when it's established (2 call legs). Scanning the variables of the 2 call legs I currentyl cannot see any relation between the job_uuid and the uuid of the resulting call legs. I may set a variable with my own unique id while originating a call, but finding the calls later on and dumping the variables fo all channels is very time consuming in terms of CPU. A workaround I tried, is to set caller-id or caller-id-number with a unique id. This works, but has the known side effects of not having a valid caller-id or caller-id-number. So my question is: Has anybody an idea how to build a relationship between job_uuid and the resulting call legs which does not require dumping the variables of all channels? Best regards Peter From solko at gcdf.pl Wed May 27 03:48:35 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 27 May 2009 12:48:35 +0200 Subject: [Freeswitch-users] Origionate a call via event_socket. relation between job_uuid and uuid In-Reply-To: <4A1D0BF2.4000800@gmx.net> References: <4A1D0BF2.4000800@gmx.net> Message-ID: <4A1D1A83.7050802@gcdf.pl> Peter P GMX pisze: > I want to do the following: > Originate a call via event_socket, I get back a job_uuid. Then I want to > control the call when it's established (2 call legs). > Scanning the variables of the 2 call legs I currentyl cannot see any > relation between the job_uuid and the uuid of the resulting call legs. > I may set a variable with my own unique id while originating a call, but > finding the calls later on and dumping the variables fo all channels is > very time consuming in terms of CPU. > > A workaround I tried, is to set caller-id or caller-id-number with a > unique id. This works, but has the known side effects of not having a > valid caller-id or caller-id-number. > > So my question is: Has anybody an idea how to build a relationship > between job_uuid and the resulting call legs which does not require > dumping the variables of all channels? > I use that scenario with my own variables set to my unique id. On event_socket I parse channel related events and then I find channel with my variable, I get channel unique id and from now I use this unique id. First I tried to uses caller-id but I need it now for the reason it is for. Szymon > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From saigop at gmail.com Wed May 27 03:53:38 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Wed, 27 May 2009 16:23:38 +0530 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> Message-ID: <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> Hi Anthony, thanks, it seems to be working, but the extension is not hanging up once I transfered the call to another mobile or to a conference. Some where I am wrong? On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > This one happens to every new guy trying to make FS into a dialer app using > JS. > > for every sessionX you create in js with the new Session constructor > > sessionX.setAutoHangup(0); > > This allows the channels to remain alive outside the script once they are > hungup/transferred etc. > > > On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: > >> Hi, >> I had some discussion with the IRC regarding the uuid_transfer gets >> hang-up where the call is originated via javascript thru event socket. I was >> suggested to install latest SVN trunk. I did that again the same issue, the >> log is attached with here http://pastebin.freeswitch.org/9103 >> >> My call flow like this, >> >> 1. api jsrun fils.js >> 2. capture the uuid >> 3. api uuid_transfer -both >> >> Both the leg gets hangedup. >> >> Someone can assist me where I am wrong. >> >> -- >> Thank you with regards, >> Gopal, >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/f50d090d/attachment.html From saigop at gmail.com Wed May 27 03:56:29 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Wed, 27 May 2009 16:26:29 +0530 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> Message-ID: <2ea4d47e0905270356u61c770b6x93d5797aec184306@mail.gmail.com> Any how let me try with uuid_kill to kill the extension uuid. On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > This one happens to every new guy trying to make FS into a dialer app using > JS. > > for every sessionX you create in js with the new Session constructor > > sessionX.setAutoHangup(0); > > This allows the channels to remain alive outside the script once they are > hungup/transferred etc. > > > On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: > >> Hi, >> I had some discussion with the IRC regarding the uuid_transfer gets >> hang-up where the call is originated via javascript thru event socket. I was >> suggested to install latest SVN trunk. I did that again the same issue, the >> log is attached with here http://pastebin.freeswitch.org/9103 >> >> My call flow like this, >> >> 1. api jsrun fils.js >> 2. capture the uuid >> 3. api uuid_transfer -both >> >> Both the leg gets hangedup. >> >> Someone can assist me where I am wrong. >> >> -- >> Thank you with regards, >> Gopal, >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/3add8cd3/attachment.html From anthony.minessale at gmail.com Wed May 27 05:50:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 May 2009 07:50:05 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> Message-ID: <191c3a030905270550s140021fbg21c3b065828e5995@mail.gmail.com> This opens the door to what you ask but to add moderator specific dtmf controls is another patch that would require specifics and most likely a bounty or contract to complete. On Tue, May 26, 2009 at 8:17 PM, Stephen Crosby wrote: > Michael, > > Pardon me for hopping on this thread, but can you explain more about this > new feature? I've been wanting something like this to apply different > behaviors for different conference members. Can this be used to provide a > 'moderator' with different behaviors bound to DTMF keys than regular > callers? > > --Stephen > > On Tue, May 26, 2009 at 5:55 PM, Michael Collins wrote: > >> >> >> On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> the easiest way would be the new feature I added to 13442 >>> >>> in the conference profile add >>> >>> >>> >>> to your >>> >>> and in your dialplan >>> >>> >>> >>> >>> or >>> >>> >>> >>> >>> Don't forget the wishlist and donate button on the main site.... >>> >>> >> And the wiki tax if you feel comfortable adding this to the wiki. If not >> let me know and I'll add it. >> -MC >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/49bd621f/attachment-0001.html From anthony.minessale at gmail.com Wed May 27 05:59:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 May 2009 07:59:42 -0500 Subject: [Freeswitch-users] Origionate a call via event_socket. relation between job_uuid and uuid In-Reply-To: <4A1D0BF2.4000800@gmx.net> References: <4A1D0BF2.4000800@gmx.net> Message-ID: <191c3a030905270559h7d4aeac4s9ab7bc58e5665df6@mail.gmail.com> Here are 3 ways: 1) subscribe to the BACKGROUND_JOB event and find the one with the same job-uuid then the body of that message is the output from your backgrounded FSAPI call which in the case of an originate will contain the uuid of the actual channel. 2) You can do as suggested and add {myvar=myval} prefix to the dialstring and look for myvar in the channel_originate event. 3) Finally you can choose the uuid in advance providing it's actually unique using: {origination_uuid=XYZ} You can use your own code to generate uuid (make sure they are unique) or ask the core to give you a new uuid with the create_uuid FSAPI call. On Wed, May 27, 2009 at 4:46 AM, Peter P GMX wrote: > I want to do the following: > Originate a call via event_socket, I get back a job_uuid. Then I want to > control the call when it's established (2 call legs). > Scanning the variables of the 2 call legs I currentyl cannot see any > relation between the job_uuid and the uuid of the resulting call legs. > I may set a variable with my own unique id while originating a call, but > finding the calls later on and dumping the variables fo all channels is > very time consuming in terms of CPU. > > A workaround I tried, is to set caller-id or caller-id-number with a > unique id. This works, but has the known side effects of not having a > valid caller-id or caller-id-number. > > So my question is: Has anybody an idea how to build a relationship > between job_uuid and the resulting call legs which does not require > dumping the variables of all channels? > > Best regards > Peter > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/5a67eace/attachment.html From anthony.minessale at gmail.com Wed May 27 06:01:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 May 2009 08:01:11 -0500 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> Message-ID: <191c3a030905270601l5d451e61y1c09afd910cb86e5@mail.gmail.com> I am not sure what you mean at this point. On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N wrote: > Hi Anthony, > thanks, it seems to be working, but the extension is not hanging up once > I transfered the call to another mobile or to a conference. Some where I am > wrong? > > On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> This one happens to every new guy trying to make FS into a dialer app >> using JS. >> >> for every sessionX you create in js with the new Session constructor >> >> sessionX.setAutoHangup(0); >> >> This allows the channels to remain alive outside the script once they are >> hungup/transferred etc. >> >> >> On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: >> >>> Hi, >>> I had some discussion with the IRC regarding the uuid_transfer gets >>> hang-up where the call is originated via javascript thru event socket. I was >>> suggested to install latest SVN trunk. I did that again the same issue, the >>> log is attached with here http://pastebin.freeswitch.org/9103 >>> >>> My call flow like this, >>> >>> 1. api jsrun fils.js >>> 2. capture the uuid >>> 3. api uuid_transfer -both >>> >>> Both the leg gets hangedup. >>> >>> Someone can assist me where I am wrong. >>> >>> -- >>> Thank you with regards, >>> Gopal, >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/505a8383/attachment.html From anthony.minessale at gmail.com Wed May 27 06:04:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 May 2009 08:04:48 -0500 Subject: [Freeswitch-users] XML config error In-Reply-To: <86a32abc0905261924y7a274f04id62d639df908fcc4@mail.gmail.com> References: <86a32abc0905261833l160cbc8fy96a3b1ee68f45a96@mail.gmail.com> <20090527021635.GA19575@jdc.jasonjgw.net> <86a32abc0905261924y7a274f04id62d639df908fcc4@mail.gmail.com> Message-ID: <191c3a030905270604x21b5429er2e54f44d593fef0b@mail.gmail.com> next time try this: 1) Read the error..... Error [unterminated ${var}] in line /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml line 12 ok, so there appears to be a problem in /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml line 12 and the problem appears to be unterminated ${var} I bet if you went and looked there it might have jumped out at you ;) I just noticed a typo in the error text while writing this email so, you can always learn from reading carefully...... On Tue, May 26, 2009 at 9:24 PM, Diego Viola wrote: > I just updated it, it was a bug that got fixed already. > > 22:19 <@bkw__> diegoviola: already fixed > 22:19 <@bkw__> update > 22:19 <@bkw__> close the } > 22:20 <@bkw__> it was a bug we fixed already this morning that catches > unclosed global preprocess vars > > Thanks, > > Diego > > On Tue, May 26, 2009 at 10:16 PM, Jason White wrote: > > Diego Viola wrote: > >> Hi, I have downloaded the latest freeswitch trunk, and when I do > >> reloadxml I get this. > >> > >> Error [unterminated ${var}] in line > >> > /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml > >> line 12 > >> > >> Any ideas? I haven't edited that file myself. > > > > Have a look in that file to see whether there's a syntax error there. If > so, > > and if it's in the source tree, fix it and submit a patch. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/a707f0d9/attachment.html From anthony.minessale at gmail.com Wed May 27 06:10:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 May 2009 08:10:37 -0500 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <006c01c9de1f$715ad360$54107a20$@net> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> <4A1C0D0E.2030301@maxpowersoft.com> <006c01c9de1f$715ad360$54107a20$@net> Message-ID: <191c3a030905270610i6ebd1e8av5891a0f38211e996@mail.gmail.com> Hey Damin! Glad to see you are still out there in the shadows. You coming back to ClueCon this year? On Tue, May 26, 2009 at 11:31 AM, Gregory Boehnlein wrote: > I can say, from having met with and talked to the CEO and founder of > Applogic that these guys are really revolutionary in their approach to cloud > computing. I spoke on a panel w/ the founder at an ISPcon several years ago, > and their approach is that of a utility company, treating computing > resources like that of a power company. Cool stuff! > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Danielson > *Sent:* Tuesday, May 26, 2009 11:39 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FS in Amazon EC2 for production? > > > > There is a cloud computing company named 3Tera (AppLogic) that does have an > international presence and will keep your FreeSWITCH instance running on a > dedicated server using Xen and HA. I spoke with one of their senior > engineers about 1 month ago in regards to actually setting up an LCR > scenario using their servers located in Europe and the United States. These > guys are a little different in the cloud computing world and I believe > closer fit the needs of a telephony application. As-is, there are companies > using 3tera for their Asterisk installs. So if you want cloud computing > with dedicated hardware resources and a set geographic location, then these > guys do it. Kind of the best of both worlds. Just a quick 2 cents... > > Regards, > Chris > > > Brian West wrote: > > > > On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote: > > > > Hey Brian, > > FreeSWITCH in EC2 is a bit of a mystery to me... > > Call me old fashioned but in my mind VoIP and geography are linked > in %99 of scenarios. Having VoIP services in a pure "cloud" > environment just doesn't sound like a good idea to me. > > Consider a "hosted" environment with clients registered to a > FreeSWITCH server. One day your instance is physically running on > hardware in Seattle. The next day it could (potentially) be running > in Chicago. That's obviously a very different routing path for your > clients. Even /if/ Amazon (or whomever) employs every routing/network > trick in the book you still won't be able to get over that change in > geography. > > > > For some people this isn't a huge difference... now if it were to swap > continents then yes it would be a problem. But I haven't seen Amazon do > this but I haven't left the instances up long enough to see. > > > > It's certainly possible a change like this may very well never > happen in practice. I wouldn't know; I've never used EC2 and I don't > even know that much about it. I'm just curious how well strictly, > practically speaking this will work in the long term. > > > > There are other companies that do this stuff but personally me... I want my > stuff running on real hardware. > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > ------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > This message has been scanned for viruses and > dangerous content by *N2Net Mailshield*, > and is > believed to be clean. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/98b7fcce/attachment-0001.html From ssa1357 at yahoo.com Wed May 27 01:26:15 2009 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Wed, 27 May 2009 01:26:15 -0700 (PDT) Subject: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal Message-ID: <148751.90414.qm@web53403.mail.re2.yahoo.com> I faced a problem when I want to compile freeswitch with mod_opal. It gives me a compilation error like bellow I used ptlib-2.6.2 and opal-3.6.2. Would someone please help me? Sincerely, Sadjad ________________________________ making all mod_opal make[5]: Entering directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' make[6]: Entering directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' Compiling mod_opal.cpp... Compiling mod_opal.cpp ... mod_opal.h:151: error: conflicting return type specified for 'virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)' /usr/include/opal/opal/localep.h:267: error: overriding 'virtual ptlib_virtual_function_changed_or_deprecated****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)' make[6]: *** [mod_opal.lo] Error 1 make[6]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' make[5]: *** [all] Error 1 make[5]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src/mod/endpoints/mod_opal' make[4]: *** [mod_opal-all] Error 1 make[4]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/root/fs/freeswitch/freeswitch.trunk/src' Making all in build make[3]: Entering directory `/root/fs/freeswitch/freeswitch.trunk/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/0c3bd872/attachment.html From shiyanov at gmail.com Wed May 27 05:38:16 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 27 May 2009 16:38:16 +0400 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function Message-ID: Hi everyone! In my environment I use FreeSwitch as media server and session border controller. SIP routing is mostly done with my private B2BUA. The problem itself is in my "hold" functionality. In details: A is calling to B: Call being answered. Then B wants to hold a call (A should hear music-on-hold, B should hear nothing but still have a call). To accomplish this I: 1. on behalf of B2BUA make call to FreeSwitch "hold" extension, get SDP for music-on-hold (moh_sdp) 2. re-invite A with SDP moh_sdp 3. re-invite B with it's own sdp and attribute 'inactive' (also tried re-invite with A's SDP and 'inactive' and with 0.0.0.0 IP in SDP and 'inactive' - same results) Result: both A and B hear music-on-hold. I suspect that the reason is in 'bridge' function.. it seems, that it still bridges media against the SIP. Here is a snippet from FreeSwitch log related to "strange" (from my point of view) behaviour - when FreeSwitch change media for B to the incorrect one. Here 1004 is B. 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2939 sofia_glue_negotiate_sdp() Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:1863 sofia_glue_tech_set_codec() Already using GSM 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2899 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2908 sofia_glue_negotiate_sdp() Set comfort noise payload to 13 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2139 sofia_glue_activate_rtp() Audio params changed for sofia/external/6782886802 at 10.249.194.9:5060 from 75.101.166.29:27426 to 75.101.166.29:22922 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/6782886802 at 10.249.194.9:5060] 10.249.194.9 port 28594 -> 75.101.166.29 port 22922 codec: 3 ms: 20 2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2165 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [75.101.166.29:22922] 2009-05-26 13:19:14 [DEBUG] sofia.c:2996 sofia_handle_sip_i_state() Channel sofia/external/1005 at 10.249.194.9 entering state [ready][200] 2009-05-26 13:19:15 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/doublenat5090/sip:1004 at 172.16.0.6 ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 Command Execute playback(local_stream://moh) EXECUTE sofia/doublenat5090/sip:1004 at 172.16.0.6 ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 playback(local_stream://moh) 2009-05-26 13:19:15 [DEBUG] mod_local_stream.c:346 local_stream_file_open() Opening Stream [moh/8000] 8000hz 2009-05-26 13:19:15 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-05-26 13:19:15 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/doublenat5090/sip:1004 at 172.16.0.6 ;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060 receive message [TRANSCODING_NECESSARY] Any thoughts would be really appreciated! Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/68e03a7b/attachment.html From brian at freeswitch.org Wed May 27 06:37:23 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 08:37:23 -0500 Subject: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal In-Reply-To: <148751.90414.qm@web53403.mail.re2.yahoo.com> References: <148751.90414.qm@web53403.mail.re2.yahoo.com> Message-ID: You have to use the SVN version of both ptlib and OPAL and it will compile. /b On May 27, 2009, at 3:26 AM, Sadjad Seyed-Ahmadian wrote: > I faced a problem when I want to compile freeswitch with mod_opal. > It gives me a compilation error like bellow > > I used ptlib-2.6.2 and opal-3.6.2. > > Would someone please help me? > > Sincerely, > Sadjad Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/a3ea9df5/attachment.html From raffaele.p.guidi at gmail.com Wed May 27 07:15:09 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Wed, 27 May 2009 16:15:09 +0200 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <3d381e170905260921k68cbec02h27297c717c2fcd91@mail.gmail.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> <3d381e170905260921k68cbec02h27297c717c2fcd91@mail.gmail.com> Message-ID: Wow, that's cool. Can you give us some figures? How many users/calls per day, what is the AMI setup, an average cost per month? Do you think it would be a feasible solution for a call center? On Tue, May 26, 2009 at 18:21, Erik Wickstrom wrote: > I've been running a production FS app on EC2 since December. It's been > really stable. Same server/instance since day1. We've haven't had any > complaints.... > > Erik > > > On Tue, May 26, 2009 at 8:11 AM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> On Tue, May 26, 2009 at 10:31 AM, Brian West >> wrote: >> > Not with FreeSWITCH in our testing. Now if you have stupid defaults >> > in your virtualization env. it might act funny but I have run FS on >> > EC2 without a problem. >> > >> > /b >> >> Hey Brian, >> >> FreeSWITCH in EC2 is a bit of a mystery to me... >> >> Call me old fashioned but in my mind VoIP and geography are linked >> in %99 of scenarios. Having VoIP services in a pure "cloud" >> environment just doesn't sound like a good idea to me. >> >> Consider a "hosted" environment with clients registered to a >> FreeSWITCH server. One day your instance is physically running on >> hardware in Seattle. The next day it could (potentially) be running >> in Chicago. That's obviously a very different routing path for your >> clients. Even /if/ Amazon (or whomever) employs every routing/network >> trick in the book you still won't be able to get over that change in >> geography. >> >> It's certainly possible a change like this may very well never >> happen in practice. I wouldn't know; I've never used EC2 and I don't >> even know that much about it. I'm just curious how well strictly, >> practically speaking this will work in the long term. >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/b1e912a1/attachment-0001.html From brian at freeswitch.org Wed May 27 07:22:07 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 09:22:07 -0500 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function In-Reply-To: References: Message-ID: Try not using RFC2543 HOLD since we do not support it. /b On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote: > Hi everyone! > > In my environment I use FreeSwitch as media server and session border > controller. SIP routing is mostly done with my private B2BUA. The > problem > itself is in my "hold" functionality. In details: A is calling to B: Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/7bfd64df/attachment.html From testeador01 at gmail.com Wed May 27 07:22:38 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 27 May 2009 09:22:38 -0500 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> Message-ID: Check the freeswitch log when the dtmf are being received, most likely your device is sending them both as Dave says, maybe the telephone you're dialing with sends it both inband and rtp. What does the log says? 2009/5/26 Drew Ozier > I've got a configuration where I receive inbound calls and dial out to a > pre-determined 800-number based on the DNIS of the call. I set application="start_dtmf"/> and have everything set up so that DTMF only > comes to me via inband. When I'm providing DTMF data to the IVR, it will > recognize a single keypress as a double-tap. My FreeSWITCH logs only contain > one DTMF entry, but when I listen to the receiving end of the call, I can > hear a hiccup in the DTMF tone that is getting played. When I do not use > 'start_dtmf', this problem goes away. I need inband DTMF detection, but I > can't have it messing up the audio stream. Any thoughts? > > -Drew Ozier > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/bb9439f9/attachment.html From shiyanov at gmail.com Wed May 27 07:34:08 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 27 May 2009 18:34:08 +0400 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function In-Reply-To: References: Message-ID: Thanks for the answer! Maybe you can advise me another scheme how to accomlish "mute" or "hold" functionality? I just wonder if mute/hold in X-Lite works with FreeSwitch.. I'll try and send the results. On Wed, May 27, 2009 at 6:22 PM, Brian West wrote: > Try not using RFC2543 HOLD since we do not support it. > /b > > On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote: > > Hi everyone! > > In my environment I use FreeSwitch as media server and session border > controller. SIP routing is mostly done with my private B2BUA. The problem > itself is in my "hold" functionality. In details: A is calling to B: > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/fbc6bbcd/attachment.html From brian at freeswitch.org Wed May 27 07:38:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 09:38:27 -0500 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function In-Reply-To: References: Message-ID: <5499232A-C58A-405A-9C5D-A891A98408D7@freeswitch.org> Yes, but if the stream says inactive we might have an issue but I can't recall off the top of my head... but Last I tested x-lite it works fine. /b On May 27, 2009, at 9:34 AM, Artem Shiyanov wrote: > Thanks for the answer! > Maybe you can advise me another scheme how to accomlish "mute" or > "hold" functionality? I just wonder if mute/hold in X-Lite works > with FreeSwitch.. I'll try and send the results. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/4887bbd1/attachment.html From drew.ozier at gmail.com Wed May 27 08:02:47 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Wed, 27 May 2009 11:02:47 -0400 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> Message-ID: <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an AudioCodes Mediant 1000 (SIP Gateway) which goes to a FreeSWITCH machine. I receive calls coming in off the T1 which goes through my Mediant 1000 which goes to my FreeSWITCH machine which makes an outbound call back through the Mediant 1000 to an IVR (bridging the inbound and outbound calls). I have (supposedly) configured the AudioCodes Mediant 1000 to pass the DTMF in-band, because I want to minimize the amount of modifications I make to the audio stream going through my system. I also want to be able to log when DTMF events occur for analysis purposes. If I do not have start_dtmf, then FreeSWITCH does not show me any DTMF in the logs, but when I have it on, it will. Here's a DTMF event in my FreeSWITCH logs: (I changed the debug message to a notice and added the callId and the nano time of the event) 2009-05-26 16:02:05 [NOTICE] switch_ivr_async.c:996 inband_dtmf_callback() 348323 61182225861200042042 at 10.1.10.10 DTMF DETECTED: 5 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1322 do_2833() Send start packet for [5] ts=1217893935 dur=160/160/2000 seq=55199 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=320/320/2000 seq=55200 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=480/480/2000 seq=55201 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=640/640/2000 seq=55202 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=800/800/2000 seq=55203 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=960/960/2000 seq=55204 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=1120/1120/2000 seq=55205 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=1280/1280/2000 seq=55206 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=1440/1440/2000 seq=55207 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=1600/1600/2000 seq=55208 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=1760/1760/2000 seq=55209 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet for [5] ts=1217893935 dur=1920/1920/2000 seq=55210 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for [5] ts=1217893935 dur=2080/2080/2000 seq=55211 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for [5] ts=1217893935 dur=2080/2080/2000 seq=55212 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for [5] ts=1217893935 dur=2080/2080/2000 seq=55213 Does the fact that switch_rtp is sending packets mean that FreeSWITCH is sending out-of-band DTMF as well as recognizing the in-band? Thanks for your quick replies, hopefully this will make my issue clearer. -Drew On Wed, May 27, 2009 at 10:22 AM, Milena wrote: > Check the freeswitch log when the dtmf are being received, most likely your > device is sending them both as Dave says, maybe the telephone you're dialing > with sends it both inband and rtp. What does the log says? > > 2009/5/26 Drew Ozier > >> I've got a configuration where I receive inbound calls and dial out to a >> pre-determined 800-number based on the DNIS of the call. I set > application="start_dtmf"/> and have everything set up so that DTMF only >> comes to me via inband. When I'm providing DTMF data to the IVR, it will >> recognize a single keypress as a double-tap. My FreeSWITCH logs only contain >> one DTMF entry, but when I listen to the receiving end of the call, I can >> hear a hiccup in the DTMF tone that is getting played. When I do not use >> 'start_dtmf', this problem goes away. I need inband DTMF detection, but I >> can't have it messing up the audio stream. Any thoughts? >> >> -Drew Ozier >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/30bc1925/attachment-0001.html From brian at freeswitch.org Wed May 27 08:12:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 10:12:12 -0500 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> Message-ID: <22D11529-2131-4AE5-9EFE-97064C4048EA@freeswitch.org> You should try to always use out of band... what is the call path because it looks like a bridged call and the far end gets the inband and the 2833 /b On May 27, 2009, at 10:02 AM, Drew Ozier wrote: > To clarify, I'm not running the IVR. I have a TDM T1 that comes in > to an AudioCodes Mediant 1000 (SIP Gateway) which goes to a > FreeSWITCH machine. I receive calls coming in off the T1 which goes > through my Mediant 1000 which goes to my FreeSWITCH machine which > makes an outbound call back through the Mediant 1000 to an IVR > (bridging the inbound and outbound calls). I have (supposedly) > configured the AudioCodes Mediant 1000 to pass the DTMF in-band, > because I want to minimize the amount of modifications I make to the > audio stream going through my system. I also want to be able to log > when DTMF events occur for analysis purposes. If I do not have > start_dtmf, then FreeSWITCH does not show me any DTMF in the logs, > but when I have it on, it will. > > Here's a DTMF event in my FreeSWITCH logs: > (I changed the debug message to a notice and added the callId and > the nano time of the event) > 2009-05-26 16:02:05 [NOTICE] switch_ivr_async.c:996 > inband_dtmf_callback() 348323 61182225861200042042 at 10.1.10.10 DTMF > DETECTED: 5 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1322 do_2833() Send start > packet for [5] ts=1217893935 dur=160/160/2000 seq=55199 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=320/320/2000 seq=55200 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=480/480/2000 seq=55201 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=640/640/2000 seq=55202 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=800/800/2000 seq=55203 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=960/960/2000 seq=55204 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=1120/1120/2000 seq=55205 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=1280/1280/2000 seq=55206 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=1440/1440/2000 seq=55207 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=1600/1600/2000 seq=55208 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=1760/1760/2000 seq=55209 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle > packet for [5] ts=1217893935 dur=1920/1920/2000 seq=55210 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end > packet for [5] ts=1217893935 dur=2080/2080/2000 seq=55211 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end > packet for [5] ts=1217893935 dur=2080/2080/2000 seq=55212 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end > packet for [5] ts=1217893935 dur=2080/2080/2000 seq=55213 > > Does the fact that switch_rtp is sending packets mean that > FreeSWITCH is sending out-of-band DTMF as well as recognizing the in- > band? > > Thanks for your quick replies, hopefully this will make my issue > clearer. > > -Drew Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/5885e66e/attachment.html From larclap at yahoo.com Wed May 27 08:16:10 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 27 May 2009 08:16:10 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> Message-ID: <002701c9dede$11452560$33cf7020$@com> Michael, Thanks for the advice. Where would you place this extension in my current conf/dialplan/default.xml to avoid the integration issue? Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the bottom of the page under 'uuid', the two hyperlinks ('ref') are broken. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 26, 2009 5:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions on build 13441 Also, in general, do the XML files in config get updated during the 'make install', or are they left as they were from the previous builds? Running "make install" or "make samples" will not overwrite your existing configuration files. NOTE: This means that when the default configuration changes you will need to update your files manually and integrate the new changes. This is one reason why I recommend making as few edits as possible to the default configuration files. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/5b683e92/attachment.html From brian at freeswitch.org Wed May 27 08:20:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 10:20:03 -0500 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <002701c9dede$11452560$33cf7020$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> Message-ID: <2BAA361D-0CE9-4809-94BB-0A6C35149B1D@freeswitch.org> Lars, Please register and correct anything you see wrong on the wiki if possible. Thanks, Brian On May 27, 2009, at 10:16 AM, Lars Zeb wrote: > Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the > bottom of the page under ?uuid?, the two hyperlinks (?ref?) are > broken. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/4fc56375/attachment-0001.html From brian at freeswitch.org Wed May 27 08:20:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 10:20:48 -0500 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <002701c9dede$11452560$33cf7020$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> Message-ID: <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> I'll get the docs section fixed this has changed and will require us to fix it. Thanks, On May 27, 2009, at 10:16 AM, Lars Zeb wrote: > Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the > bottom of the page under ?uuid?, the two hyperlinks (?ref?) are > broken. > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/2a5741f6/attachment.html From drew.ozier at gmail.com Wed May 27 08:25:53 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Wed, 27 May 2009 11:25:53 -0400 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <22D11529-2131-4AE5-9EFE-97064C4048EA@freeswitch.org> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> <22D11529-2131-4AE5-9EFE-97064C4048EA@freeswitch.org> Message-ID: <2388e50e0905270825w2075b287hfec5ee623e4520ea@mail.gmail.com> Hey Brian, It is a bridged call. Here's the majority of my dialplan: On Wed, May 27, 2009 at 11:12 AM, Brian West wrote: > You should try to always use out of band... what is the call path because > it looks like a bridged call and the far end gets the inband and the 2833 > /b > > On May 27, 2009, at 10:02 AM, Drew Ozier wrote: > > To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an > AudioCodes Mediant 1000 (SIP Gateway) which goes to a FreeSWITCH machine. I > receive calls coming in off the T1 which goes through my Mediant 1000 which > goes to my FreeSWITCH machine which makes an outbound call back through the > Mediant 1000 to an IVR (bridging the inbound and outbound calls). I have > (supposedly) configured the AudioCodes Mediant 1000 to pass the DTMF > in-band, because I want to minimize the amount of modifications I make to > the audio stream going through my system. I also want to be able to log when > DTMF events occur for analysis purposes. If I do not have start_dtmf, then > FreeSWITCH does not show me any DTMF in the logs, but when I have it on, it > will. > > Here's a DTMF event in my FreeSWITCH logs: > (I changed the debug message to a notice and added the callId and the nano > time of the event) > 2009-05-26 16:02:05 [NOTICE] switch_ivr_async.c:996 inband_dtmf_callback() > 348323 61182225861200042042 at 10.1.10.10 DTMF DETECTED: 5 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1322 do_2833() Send start packet > for [5] ts=1217893935 dur=160/160/2000 seq=55199 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=320/320/2000 seq=55200 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=480/480/2000 seq=55201 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=640/640/2000 seq=55202 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=800/800/2000 seq=55203 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=960/960/2000 seq=55204 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1120/1120/2000 seq=55205 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1280/1280/2000 seq=55206 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1440/1440/2000 seq=55207 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1600/1600/2000 seq=55208 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1760/1760/2000 seq=55209 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1920/1920/2000 seq=55210 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for > [5] ts=1217893935 dur=2080/2080/2000 seq=55211 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for > [5] ts=1217893935 dur=2080/2080/2000 seq=55212 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for > [5] ts=1217893935 dur=2080/2080/2000 seq=55213 > > Does the fact that switch_rtp is sending packets mean that FreeSWITCH is > sending out-of-band DTMF as well as recognizing the in-band? > > Thanks for your quick replies, hopefully this will make my issue clearer. > > -Drew > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/18137f95/attachment.html From damin at nacs.net Wed May 27 08:29:01 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Wed, 27 May 2009 11:29:01 -0400 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <191c3a030905270610i6ebd1e8av5891a0f38211e996@mail.gmail.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> <4A1C0D0E.2030301@maxpowersoft.com> <006c01c9de1f$715ad360$54107a20$@net> <191c3a030905270610i6ebd1e8av5891a0f38211e996@mail.gmail.com> Message-ID: <031d01c9dedf$dc39d260$94ad7720$@net> Still here.. working.. making a living.. you know the drill. Not sure about ClueCon this year.. I've curtailed a lot of my traveling and talking this year as I've been doing a lot more consulting work. I'll look into it.. I may be able to drive up if I can spare the time.. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, May 27, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in Amazon EC2 for production? Hey Damin! Glad to see you are still out there in the shadows. You coming back to ClueCon this year? On Tue, May 26, 2009 at 11:31 AM, Gregory Boehnlein wrote: I can say, from having met with and talked to the CEO and founder of Applogic that these guys are really revolutionary in their approach to cloud computing. I spoke on a panel w/ the founder at an ISPcon several years ago, and their approach is that of a utility company, treating computing resources like that of a power company. Cool stuff! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Danielson Sent: Tuesday, May 26, 2009 11:39 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in Amazon EC2 for production? There is a cloud computing company named 3Tera (AppLogic) that does have an international presence and will keep your FreeSWITCH instance running on a dedicated server using Xen and HA. I spoke with one of their senior engineers about 1 month ago in regards to actually setting up an LCR scenario using their servers located in Europe and the United States. These guys are a little different in the cloud computing world and I believe closer fit the needs of a telephony application. As-is, there are companies using 3tera for their Asterisk installs. So if you want cloud computing with dedicated hardware resources and a set geographic location, then these guys do it. Kind of the best of both worlds. Just a quick 2 cents... Regards, Chris Brian West wrote: On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote: Hey Brian, FreeSWITCH in EC2 is a bit of a mystery to me... Call me old fashioned but in my mind VoIP and geography are linked in %99 of scenarios. Having VoIP services in a pure "cloud" environment just doesn't sound like a good idea to me. Consider a "hosted" environment with clients registered to a FreeSWITCH server. One day your instance is physically running on hardware in Seattle. The next day it could (potentially) be running in Chicago. That's obviously a very different routing path for your clients. Even /if/ Amazon (or whomever) employs every routing/network trick in the book you still won't be able to get over that change in geography. For some people this isn't a huge difference... now if it were to swap continents then yes it would be a problem. But I haven't seen Amazon do this but I haven't left the instances up long enough to see. It's certainly possible a change like this may very well never happen in practice. I wouldn't know; I've never used EC2 and I don't even know that much about it. I'm just curious how well strictly, practically speaking this will work in the long term. There are other companies that do this stuff but personally me... I want my stuff running on real hardware. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _____ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- This message has been scanned for viruses and dangerous content by N2Net Mailshield, and is believed to be clean. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -- This message has been scanned for viruses and dangerous content by N2Net Mailshield , and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/9fd92388/attachment-0001.html From brian at freeswitch.org Wed May 27 08:30:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 10:30:01 -0500 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <2388e50e0905270825w2075b287hfec5ee623e4520ea@mail.gmail.com> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> <22D11529-2131-4AE5-9EFE-97064C4048EA@freeswitch.org> <2388e50e0905270825w2075b287hfec5ee623e4520ea@mail.gmail.com> Message-ID: <8A2418F7-45F8-46A3-94DD-4DD426DBF7A0@freeswitch.org> This is what starts the inband detector.. what is the far side? /b On May 27, 2009, at 10:25 AM, Drew Ozier wrote: > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/c63cc0c4/attachment.html From drew.ozier at gmail.com Wed May 27 08:45:56 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Wed, 27 May 2009 11:45:56 -0400 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <8A2418F7-45F8-46A3-94DD-4DD426DBF7A0@freeswitch.org> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> <22D11529-2131-4AE5-9EFE-97064C4048EA@freeswitch.org> <2388e50e0905270825w2075b287hfec5ee623e4520ea@mail.gmail.com> <8A2418F7-45F8-46A3-94DD-4DD426DBF7A0@freeswitch.org> Message-ID: <2388e50e0905270845u5b5b87b8p50b2fa7aef1f5b55@mail.gmail.com> I'm sorry, far side meaning what exactly? The call flow is as follows: TDM T1 -> AudioCodes Mediant 1000 -> FreeSWITCH -> AudioCodes Mediant 1000 -> TDM T1 FreeSWITCH gets an inbound SIP call and makes an outbound SIP call back to the Mediant 1000, bridging the two calls. The Mediant 1000 sends the call back out the T1 and that goes to my customer's IVR. I have no control over anything after it leaves my T1. Is there some other piece of information I'm missing? Do you want my sip_profiles? -Drew On Wed, May 27, 2009 at 11:30 AM, Brian West wrote: > This is what starts the inband detector.. what is the far side? > > /b > > > On May 27, 2009, at 10:25 AM, Drew Ozier wrote: > > > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/552a4b98/attachment.html From testeador01 at gmail.com Wed May 27 08:49:28 2009 From: testeador01 at gmail.com (Milena) Date: Wed, 27 May 2009 10:49:28 -0500 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> Message-ID: Are the configurations of the audiocodes set to send only in-audio dtmf? make sure everything is sending only in-audio 2009/5/27 Drew Ozier > To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an > AudioCodes Mediant 1000 (SIP Gateway) which goes to a FreeSWITCH machine. I > receive calls coming in off the T1 which goes through my Mediant 1000 which > goes to my FreeSWITCH machine which makes an outbound call back through the > Mediant 1000 to an IVR (bridging the inbound and outbound calls). I have > (supposedly) configured the AudioCodes Mediant 1000 to pass the DTMF > in-band, because I want to minimize the amount of modifications I make to > the audio stream going through my system. I also want to be able to log when > DTMF events occur for analysis purposes. If I do not have start_dtmf, then > FreeSWITCH does not show me any DTMF in the logs, but when I have it on, it > will. > > Here's a DTMF event in my FreeSWITCH logs: > (I changed the debug message to a notice and added the callId and the nano > time of the event) > 2009-05-26 16:02:05 [NOTICE] switch_ivr_async.c:996 inband_dtmf_callback() > 348323 61182225861200042042 at 10.1.10.10 DTMF DETECTED: 5 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1322 do_2833() Send start packet > for [5] ts=1217893935 dur=160/160/2000 seq=55199 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=320/320/2000 seq=55200 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=480/480/2000 seq=55201 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=640/640/2000 seq=55202 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=800/800/2000 seq=55203 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=960/960/2000 seq=55204 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1120/1120/2000 seq=55205 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1280/1280/2000 seq=55206 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1440/1440/2000 seq=55207 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1600/1600/2000 seq=55208 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1760/1760/2000 seq=55209 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send middle packet > for [5] ts=1217893935 dur=1920/1920/2000 seq=55210 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for > [5] ts=1217893935 dur=2080/2080/2000 seq=55211 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for > [5] ts=1217893935 dur=2080/2080/2000 seq=55212 > 2009-05-26 16:02:05 [DEBUG] switch_rtp.c:1258 do_2833() Send end packet for > [5] ts=1217893935 dur=2080/2080/2000 seq=55213 > > Does the fact that switch_rtp is sending packets mean that FreeSWITCH is > sending out-of-band DTMF as well as recognizing the in-band? > > Thanks for your quick replies, hopefully this will make my issue clearer. > > -Drew > > > On Wed, May 27, 2009 at 10:22 AM, Milena wrote: > >> Check the freeswitch log when the dtmf are being received, most likely >> your device is sending them both as Dave says, maybe the telephone you're >> dialing with sends it both inband and rtp. What does the log says? >> >> 2009/5/26 Drew Ozier >> >>> I've got a configuration where I receive inbound calls and dial out to a >>> pre-determined 800-number based on the DNIS of the call. I set >> application="start_dtmf"/> and have everything set up so that DTMF only >>> comes to me via inband. When I'm providing DTMF data to the IVR, it will >>> recognize a single keypress as a double-tap. My FreeSWITCH logs only contain >>> one DTMF entry, but when I listen to the receiving end of the call, I can >>> hear a hiccup in the DTMF tone that is getting played. When I do not use >>> 'start_dtmf', this problem goes away. I need inband DTMF detection, but I >>> can't have it messing up the audio stream. Any thoughts? >>> >>> -Drew Ozier >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/6802471c/attachment.html From brian at freeswitch.org Wed May 27 08:50:22 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 10:50:22 -0500 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <2388e50e0905270845u5b5b87b8p50b2fa7aef1f5b55@mail.gmail.com> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> <22D11529-2131-4AE5-9EFE-97064C4048EA@freeswitch.org> <2388e50e0905270825w2075b287hfec5ee623e4520ea@mail.gmail.com> <8A2418F7-45F8-46A3-94DD-4DD426DBF7A0@freeswitch.org> <2388e50e0905270845u5b5b87b8p50b2fa7aef1f5b55@mail.gmail.com> Message-ID: I would configure both mediant 1000's to do rfc2833, or disable 2833 on your sofia profile and do 100% inband.. right now I have a feeling the mediant is getting the inband and the 2833 causing your problem. /b On May 27, 2009, at 10:45 AM, Drew Ozier wrote: > I'm sorry, far side meaning what exactly? The call flow is as follows: > TDM T1 -> AudioCodes Mediant 1000 -> FreeSWITCH -> AudioCodes > Mediant 1000 -> TDM T1 > FreeSWITCH gets an inbound SIP call and makes an outbound SIP call > back to the Mediant 1000, bridging the two calls. The Mediant 1000 > sends the call back out the T1 and that goes to my customer's IVR. I > have no control over anything after it leaves my T1. > Is there some other piece of information I'm missing? Do you want my > sip_profiles? > > -Drew Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/0e9982c0/attachment-0001.html From brian at freeswitch.org Wed May 27 08:52:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 10:52:50 -0500 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> Message-ID: <247344A9-0FE1-4964-8365-34874C194432@freeswitch.org> I highly recommend you DO NOT use inband AT ALL if possible. out of band is king. /b On May 27, 2009, at 10:49 AM, Milena wrote: > Are the configurations of the audiocodes set to send only in-audio > dtmf? make sure everything is sending only in-audio Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/2e9b7850/attachment.html From drew.ozier at gmail.com Wed May 27 09:11:26 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Wed, 27 May 2009 12:11:26 -0400 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <247344A9-0FE1-4964-8365-34874C194432@freeswitch.org> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> <247344A9-0FE1-4964-8365-34874C194432@freeswitch.org> Message-ID: <2388e50e0905270911u7b009f9fy4f7d7abfc7b781e9@mail.gmail.com> The AudioCodes machines are always a bit of a nuisance to get acting the way I want, but I think I have them sending all DTMF in-band. How do I disable 2833 in my sofia profile? -Drew On Wed, May 27, 2009 at 11:52 AM, Brian West wrote: > I highly recommend you DO NOT use inband AT ALL if possible. out of band > is king. > /b > > On May 27, 2009, at 10:49 AM, Milena wrote: > > Are the configurations of the audiocodes set to send only in-audio dtmf? > make sure everything is sending only in-audio > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/eab39109/attachment.html From Prometheus001 at gmx.net Wed May 27 09:13:33 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 27 May 2009 18:13:33 +0200 Subject: [Freeswitch-users] Origionate a call via event_socket. relation between job_uuid and uuid In-Reply-To: <191c3a030905270559h7d4aeac4s9ab7bc58e5665df6@mail.gmail.com> References: <4A1D0BF2.4000800@gmx.net> <191c3a030905270559h7d4aeac4s9ab7bc58e5665df6@mail.gmail.com> Message-ID: <4A1D66AD.9050807@gmx.net> Hello Thanks for your hints, I now added {initiator_uuid=my_uuid} prefix to the dialstring. Then I catch the channel_answer event, get this variable_initiator_uuid and pass it to the application. This works like a charm. Thanks to all. Best regards Peter Anthony Minessale schrieb: > Here are 3 ways: > > 1) subscribe to the BACKGROUND_JOB event and find the one with the > same job-uuid > then the body of that message is the output from your backgrounded > FSAPI call which in the case > of an originate will contain the uuid of the actual channel. > > 2) You can do as suggested and add {myvar=myval} prefix to the > dialstring and look for > myvar in the channel_originate event. > > 3) Finally you can choose the uuid in advance providing it's actually > unique using: > > {origination_uuid=XYZ} > > You can use your own code to generate uuid (make sure they are > unique) or > ask the core to give you a new uuid with the create_uuid FSAPI call. > > > > > On Wed, May 27, 2009 at 4:46 AM, Peter P GMX > wrote: > > I want to do the following: > Originate a call via event_socket, I get back a job_uuid. Then I > want to > control the call when it's established (2 call legs). > Scanning the variables of the 2 call legs I currentyl cannot see any > relation between the job_uuid and the uuid of the resulting call legs. > I may set a variable with my own unique id while originating a > call, but > finding the calls later on and dumping the variables fo all > channels is > very time consuming in terms of CPU. > > A workaround I tried, is to set caller-id or caller-id-number with a > unique id. This works, but has the known side effects of not having a > valid caller-id or caller-id-number. > > So my question is: Has anybody an idea how to build a relationship > between job_uuid and the resulting call legs which does not require > dumping the variables of all channels? > > Best regards > Peter > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From erik at erikwickstrom.com Wed May 27 09:18:45 2009 From: erik at erikwickstrom.com (Erik Wickstrom) Date: Wed, 27 May 2009 09:18:45 -0700 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> <3d381e170905260921k68cbec02h27297c717c2fcd91@mail.gmail.com> Message-ID: <3d381e170905270918h101a5d3at9240ed5d56c8a879@mail.gmail.com> It's been under pretty light use. About 20 users. A bunch of DIDs coming in and some outbound campaigns. A couple hundred calls a day. (we also did a test for an outbound campaign with 8 telemarketers making 1000s of calls in a day -- worked great!) The AMI is based on Ubuntu 8.04. We're using the smallest instance at this point, so it's $70/mo + bandwidth/storage ~~ maybe $80/mo. The loadavg is always at 0.00. Erik On Wed, May 27, 2009 at 7:15 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > Wow, that's cool. Can you give us some figures? How many users/calls per > day, what is the AMI setup, an average cost per month? Do you think it would > be a feasible solution for a call center? > > > On Tue, May 26, 2009 at 18:21, Erik Wickstrom wrote: > >> I've been running a production FS app on EC2 since December. It's been >> really stable. Same server/instance since day1. We've haven't had any >> complaints.... >> >> Erik >> >> >> On Tue, May 26, 2009 at 8:11 AM, Kristian Kielhofner < >> kristian.kielhofner at gmail.com> wrote: >> >>> On Tue, May 26, 2009 at 10:31 AM, Brian West >>> wrote: >>> > Not with FreeSWITCH in our testing. Now if you have stupid defaults >>> > in your virtualization env. it might act funny but I have run FS on >>> > EC2 without a problem. >>> > >>> > /b >>> >>> Hey Brian, >>> >>> FreeSWITCH in EC2 is a bit of a mystery to me... >>> >>> Call me old fashioned but in my mind VoIP and geography are linked >>> in %99 of scenarios. Having VoIP services in a pure "cloud" >>> environment just doesn't sound like a good idea to me. >>> >>> Consider a "hosted" environment with clients registered to a >>> FreeSWITCH server. One day your instance is physically running on >>> hardware in Seattle. The next day it could (potentially) be running >>> in Chicago. That's obviously a very different routing path for your >>> clients. Even /if/ Amazon (or whomever) employs every routing/network >>> trick in the book you still won't be able to get over that change in >>> geography. >>> >>> It's certainly possible a change like this may very well never >>> happen in practice. I wouldn't know; I've never used EC2 and I don't >>> even know that much about it. I'm just curious how well strictly, >>> practically speaking this will work in the long term. >>> >>> -- >>> Kristian Kielhofner >>> http://www.astlinux.org >>> http://blog.krisk.org >>> http://www.star2star.com >>> http://www.submityoursip.com >>> http://www.voalte.com >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/cd7c3a6c/attachment.html From gregt at cgicommunications.com Wed May 27 09:25:45 2009 From: gregt at cgicommunications.com (Greg Thoen) Date: Wed, 27 May 2009 12:25:45 -0400 Subject: [Freeswitch-users] Transcoding question Message-ID: Is transcoding a wav file something that I should try to avoid? I have my files encoded as 16kbit, 8000hz, mono files, yet still in the logs I see: 2009-05-27 11:55:18 [DEBUG] switch_ivr_play_say.c:1084 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-05-27 11:55:18 [DEBUG] switch_core_io.c:649 switch_core_session_write_frame() sofia/internal/ 5853297721 at 64.24.35.78 receive message [TRANSCODING_NECESSARY] I have to think that not transcoding would be easier on the server, yet I don't know why it needs to transcode the audio file if it is, I think, already in the correct format for that codec... -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 http://www.cgicommunications.com/ http://www.elocallink.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/29434144/attachment-0001.html From stevecrozz at gmail.com Wed May 27 09:27:00 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Wed, 27 May 2009 09:27:00 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <191c3a030905270550s140021fbg21c3b065828e5995@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> <191c3a030905270550s140021fbg21c3b065828e5995@mail.gmail.com> Message-ID: <11990ade0905270927o4d0a15edm750d46aef1c9bc0@mail.gmail.com> Thanks j3flight, I have used that method, but the profiles seem to be for conferences not users. So I give a conference a profile, and everybody in the conference shares the profile settings. If I'm wrong, let me know, otherwise I'll stop hijacking this thread. --Stephen On Wed, May 27, 2009 at 5:50 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > This opens the door to what you ask but to add moderator specific dtmf > controls is another patch that would require specifics and most likely a > bounty or contract to complete. > > > > On Tue, May 26, 2009 at 8:17 PM, Stephen Crosby wrote: > >> Michael, >> >> Pardon me for hopping on this thread, but can you explain more about this >> new feature? I've been wanting something like this to apply different >> behaviors for different conference members. Can this be used to provide a >> 'moderator' with different behaviors bound to DTMF keys than regular >> callers? >> >> --Stephen >> >> On Tue, May 26, 2009 at 5:55 PM, Michael Collins wrote: >> >>> >>> >>> On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> the easiest way would be the new feature I added to 13442 >>>> >>>> in the conference profile add >>>> >>>> >>>> >>>> to your >>>> >>>> and in your dialplan >>>> >>>> >>>> >>>> >>>> or >>>> >>>> >>>> >>>> >>>> Don't forget the wishlist and donate button on the main site.... >>>> >>>> >>> And the wiki tax if you feel comfortable adding this to the wiki. If not >>> let me know and I'll add it. >>> -MC >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/acf4c793/attachment.html From diego.viola at gmail.com Wed May 27 09:32:14 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 27 May 2009 12:32:14 -0400 Subject: [Freeswitch-users] XML config error In-Reply-To: <191c3a030905270604x21b5429er2e54f44d593fef0b@mail.gmail.com> References: <86a32abc0905261833l160cbc8fy96a3b1ee68f45a96@mail.gmail.com> <20090527021635.GA19575@jdc.jasonjgw.net> <86a32abc0905261924y7a274f04id62d639df908fcc4@mail.gmail.com> <191c3a030905270604x21b5429er2e54f44d593fef0b@mail.gmail.com> Message-ID: <86a32abc0905270932t71bef2e8i324f84cb67b6d4d0@mail.gmail.com> Thanks Anthony, Will do that next time :) On Wed, May 27, 2009 at 9:04 AM, Anthony Minessale wrote: > next time try this: > > 1) Read the error..... > > Error [unterminated ${var}] in line > /usr/local/freeswitch/conf/ > autoload_configs/../jingle_profiles/client.xml > line 12 > > ok, so there appears to be a problem in > /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml > line 12 > > and the problem appears to be unterminated ${var} > > I bet if you went and looked there it might have jumped out at you ;) > I just noticed a typo in the error text while writing this email so, you can > always learn from reading carefully...... > > > > On Tue, May 26, 2009 at 9:24 PM, Diego Viola wrote: >> >> I just updated it, it was a bug that got fixed already. >> >> 22:19 <@bkw__> diegoviola: already fixed >> 22:19 <@bkw__> update >> 22:19 <@bkw__> close the } >> 22:20 <@bkw__> it was a bug we fixed already this morning that catches >> unclosed global preprocess vars >> >> Thanks, >> >> Diego >> >> On Tue, May 26, 2009 at 10:16 PM, Jason White wrote: >> > Diego Viola wrote: >> >> Hi, I have downloaded the latest freeswitch trunk, and when I do >> >> reloadxml I get this. >> >> >> >> Error [unterminated ${var}] in line >> >> >> >> /usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml >> >> line 12 >> >> >> >> Any ideas? I haven't edited that file myself. >> > >> > Have a look in that file to see whether there's a syntax error there. If >> > so, >> > and if it's in the source tree, fix it and submit a patch. >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed May 27 09:33:06 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 11:33:06 -0500 Subject: [Freeswitch-users] Transcoding question In-Reply-To: References: Message-ID: <06D21F49-497C-4B12-AB30-845C168D7CF1@freeswitch.org> Chances are those wav files are L16 either way its not an issue to transcode those... now if you were transcoding 48k to 8k all day long then I would worry but I think you're looking for savings in places where it doesn't matter in FreeSWITCH. /b On May 27, 2009, at 11:25 AM, Greg Thoen wrote: > Is transcoding a wav file something that I should try to avoid? I > have my files encoded as 16kbit, 8000hz, mono files, yet still in > the logs I see: Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/e2b0cce6/attachment.html From mariusz_kolo at wp.pl Wed May 27 09:40:08 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk_WP?=) Date: Wed, 27 May 2009 18:40:08 +0200 Subject: [Freeswitch-users] AutoChanging port problem Message-ID: <4A1D6CE8.2060809@wp.pl> Hello I have about 100 phones in lan. Every phone have set rtp port to 20000 (very simple phone 8Level you can set one rtp port not range like linsys can) my logs in freeswithc looks: 2009-05-27 17:51:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 101->166 in context default 2009-05-27 17:51:10 [NOTICE] mod_dptools.c:1274 system_session_function() Executing command: mkdir -p /records/2009-05-27 2009-05-27 17:51:10 [INFO] switch_core_session.c:1391 switch_core_session_execute_application() Sending early media 2009-05-27 17:51:10 [INFO] mod_sofia.c:1448 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1243416430 1243416431 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0 m=audio 23040 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv 2009-05-27 17:51:10 [NOTICE] mod_sofia.c:1451 sofia_receive_message() Pre-Answer sofia/internal/101 at 192.168.1.1! 2009-05-27 17:51:10 [NOTICE] switch_channel.c:602 switch_channel_set_name() New Channel sofia/internal/sip:166 at 192.168.1.166:5060;fs_nat=yes;fs_path=sip%3A166%40192.168.1.166%3A5060 [32e3b290-4ad6-11de-981e-f9f8384b4b12] 2009-05-27 17:51:10 [NOTICE] sofia.c:3101 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:166 at 192.168.1.166:5060;fs_nat=yes;fs_path=sip%3A166%40192.168.1.166%3A5060! 2009-05-27 17:51:10 [INFO] switch_rtp.c:1800 rtp_common_read() Auto Changing port from 192.168.1.101:16384 to 192.168.1.101:1045 Phone has set rtp port to 16384 but freeswitch changes to 1045 (rtp range on freeswitch is default from 16384 to 32768). Why has it change to 1045 There isn't any firewall on this server Can be a problem if every phone in net has set in his config the same rtp port ? freeSwitch ver: FreeSWITCH Version 1.0.trunk (13430) linux: Linux pbx 2.6.24-23-server #1 SMP Mon Jan 26 00:55:21 UTC 2009 i686 GNU/Linux Thanks From jcromes at gmail.com Wed May 27 09:45:00 2009 From: jcromes at gmail.com (j3flight) Date: Wed, 27 May 2009 09:45:00 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <11990ade0905270927o4d0a15edm750d46aef1c9bc0@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> <191c3a030905270550s140021fbg21c3b065828e5995@mail.gmail.com> <11990ade0905270927o4d0a15edm750d46aef1c9bc0@mail.gmail.com> Message-ID: <23746348.post@talk.nabble.com> Nope, it seems you are absolutley correct. I had setup my conferences this way, but hadn't experimented yet. I tried putting two users into the same conference using different profiles, but they both had the same caller controls. Bummer. I believe we can get around this though, assuming I can call javascript from the caller controls. Currently, I am using a trick in my conference IVR where I scan the UUIDs in a conference looking for an IsModerator variable that I set when they were dropped in. If a user has that variable set to 1, they are a moderator. I could use that same trick from the caller controls to determine whether that user has moderator permissions and can act (or not act) on their button press. I'll give it a shot at some point and post back. Stephen Crosby-2 wrote: > > Thanks j3flight, I have used that method, but the profiles seem to be for > conferences not users. So I give a conference a profile, and everybody in > the conference shares the profile settings. If I'm wrong, let me know, > otherwise I'll stop hijacking this thread. > > --Stephen > > On Wed, May 27, 2009 at 5:50 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> This opens the door to what you ask but to add moderator specific dtmf >> controls is another patch that would require specifics and most likely a >> bounty or contract to complete. >> >> >> >> On Tue, May 26, 2009 at 8:17 PM, Stephen Crosby >> wrote: >> >>> Michael, >>> >>> Pardon me for hopping on this thread, but can you explain more about >>> this >>> new feature? I've been wanting something like this to apply different >>> behaviors for different conference members. Can this be used to provide >>> a >>> 'moderator' with different behaviors bound to DTMF keys than regular >>> callers? >>> >>> --Stephen >>> >>> On Tue, May 26, 2009 at 5:55 PM, Michael Collins >>> wrote: >>> >>>> >>>> >>>> On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> the easiest way would be the new feature I added to 13442 >>>>> >>>>> in the conference profile add >>>>> >>>>> >>>>> >>>>> to your >>>>> >>>>> and in your dialplan >>>>> >>>>> >>>>> >>>>> >>>>> or >>>>> >>>>> >>>>> >>>>> >>>>> Don't forget the wishlist and donate button on the main site.... >>>>> >>>>> >>>> And the wiki tax if you feel comfortable adding this to the wiki. If >>>> not >>>> let me know and I'll add it. >>>> -MC >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23746348.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From shiyanov at gmail.com Wed May 27 09:45:26 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 27 May 2009 20:45:26 +0400 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function In-Reply-To: References: Message-ID: I've just tried native "hold" X-Lite hold with FreeSwitch - it works. When you hold call, X-Lite (eyeBeam) sends re-INVITE with SDP v=0 o=- 8 3 IN IP4 172.16.0.6 s=CounterPath eyeBeam 1.5 c=IN IP4 0.0.0.0 t=0 0 m=audio 43362 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendonly SJPhone does the same way. I also tried to send the same - own SDP with c=IN IP4 0.0.0.0 a=sendonly but no luck. Probably mentioned sopftphones simply do not play incomming media when the call is holded? In general, is how should I hold a call using FS? On Wed, May 27, 2009 at 6:34 PM, Artem Shiyanov wrote: > Thanks for the answer! > Maybe you can advise me another scheme how to accomlish "mute" or "hold" > functionality? I just wonder if mute/hold in X-Lite works with FreeSwitch.. > I'll try and send the results. > > > > On Wed, May 27, 2009 at 6:22 PM, Brian West wrote: > >> Try not using RFC2543 HOLD since we do not support it. >> /b >> >> On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote: >> >> Hi everyone! >> >> In my environment I use FreeSwitch as media server and session border >> controller. SIP routing is mostly done with my private B2BUA. The problem >> itself is in my "hold" functionality. In details: A is calling to B: >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/d36ae98e/attachment.html From brian at freeswitch.org Wed May 27 09:47:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 11:47:21 -0500 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function In-Reply-To: References: Message-ID: Yes you won't hear anything if you press HOLD... the other caller you were talking to will hear music. /b On May 27, 2009, at 11:45 AM, Artem Shiyanov wrote: > Probably mentioned sopftphones simply do not play incomming media > when the call is holded? > In general, is how should I hold a call using FS? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/db7adc4e/attachment.html From msc at freeswitch.org Wed May 27 10:04:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 May 2009 10:04:55 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> Message-ID: <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> On Wed, May 27, 2009 at 8:20 AM, Brian West wrote: > I'll get the docs section fixed this has changed and will require us to fix > it. > Thanks, > > On May 27, 2009, at 10:16 AM, Lars Zeb wrote: > > Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the bottom of > the page under ?uuid?, the two hyperlinks (?ref?) are broken. > > > Fixed. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/86044c65/attachment.html From shiyanov at gmail.com Wed May 27 10:08:35 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 27 May 2009 21:08:35 +0400 Subject: [Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function In-Reply-To: References: Message-ID: I've checked again (thanks for your hint) and, really, FreeSwitch is so wise that he plays MOH for the holded person and and silence for the hold initiator! I'm going to is this work_flow for my B2BUA. Thanks! On Wed, May 27, 2009 at 8:47 PM, Brian West wrote: > Yes you won't hear anything if you press HOLD... the other caller you were > talking to will hear music. > /b > > On May 27, 2009, at 11:45 AM, Artem Shiyanov wrote: > > Probably mentioned sopftphones simply do not play incomming media when the > call is holded? > In general, is how should I hold a call using FS? > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/e2b56320/attachment.html From larclap at yahoo.com Wed May 27 11:28:28 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 27 May 2009 11:28:28 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> Message-ID: <00c601c9def8$ee5eb690$cb1c23b0$@com> Michael, thanks for fixing the links. And what about your advice on defining an extension that pertains all internal extensions (1000-1019) which I have currently put in conf/dialplan/default.xml? Is there an alternative file I should be putting this into? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 27, 2009 10:05 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions on build 13441 On Wed, May 27, 2009 at 8:20 AM, Brian West wrote: I'll get the docs section fixed this has changed and will require us to fix it. Thanks, On May 27, 2009, at 10:16 AM, Lars Zeb wrote: Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the bottom of the page under 'uuid', the two hyperlinks ('ref') are broken. Fixed. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/3fc15c9c/attachment-0001.html From msc at freeswitch.org Wed May 27 11:55:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 May 2009 11:55:57 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <00c601c9def8$ee5eb690$cb1c23b0$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> <00c601c9def8$ee5eb690$cb1c23b0$@com> Message-ID: <87f2f3b90905271155t3bcdc087g986066949580dac7@mail.gmail.com> On Wed, May 27, 2009 at 11:28 AM, Lars Zeb wrote: > Michael, thanks for fixing the links. > > > > And what about your advice on defining an extension that pertains all > internal extensions (1000-1019) which I have currently put in > conf/dialplan/default.xml? Is there an alternative file I should be putting > this into? > > If you need to change the default local_extensions dp entry then there's not a whole lot of options. You could manually delete (or carefully comment out) all of the default stuff and then create your own custom version of that extension and place it in conf/dialplan/default/00_MyExtension.xml or something like that... The reason I say carefully comment out is that you cannot nest comments in XML. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/d108d756/attachment.html From raffaele.p.guidi at gmail.com Wed May 27 12:08:00 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Wed, 27 May 2009 21:08:00 +0200 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <3d381e170905270918h101a5d3at9240ed5d56c8a879@mail.gmail.com> References: <011301c9dd84$c3053610$490fa230$@com> <4A1B2D26.90202@gmx.net> <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> <2d9149cd0905260811w52a0e00ap1b362cbcb4e17b91@mail.gmail.com> <3d381e170905260921k68cbec02h27297c717c2fcd91@mail.gmail.com> <3d381e170905270918h101a5d3at9240ed5d56c8a879@mail.gmail.com> Message-ID: Well actually I have an average 15 telemarketers running on a small (650 euros) server with the same load (an average 1%). Of course availability and scalability are on a different level but it's no easy to build a case - which is the more cost effective scenario with this and a growing (50+ operators) load? Things to think about. Regards, Raffaele On Wed, May 27, 2009 at 18:18, Erik Wickstrom wrote: > It's been under pretty light use. About 20 users. A bunch of DIDs coming > in and some outbound campaigns. A couple hundred calls a day. (we also did > a test for an outbound campaign with 8 telemarketers making 1000s of calls > in a day -- worked great!) > > The AMI is based on Ubuntu 8.04. We're using the smallest instance at this > point, so it's $70/mo + bandwidth/storage ~~ maybe $80/mo. > > The loadavg is always at 0.00. > > Erik > > > On Wed, May 27, 2009 at 7:15 AM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > >> Wow, that's cool. Can you give us some figures? How many users/calls per >> day, what is the AMI setup, an average cost per month? Do you think it would >> be a feasible solution for a call center? >> >> >> On Tue, May 26, 2009 at 18:21, Erik Wickstrom wrote: >> >>> I've been running a production FS app on EC2 since December. It's been >>> really stable. Same server/instance since day1. We've haven't had any >>> complaints.... >>> >>> Erik >>> >>> >>> On Tue, May 26, 2009 at 8:11 AM, Kristian Kielhofner < >>> kristian.kielhofner at gmail.com> wrote: >>> >>>> On Tue, May 26, 2009 at 10:31 AM, Brian West >>>> wrote: >>>> > Not with FreeSWITCH in our testing. Now if you have stupid defaults >>>> > in your virtualization env. it might act funny but I have run FS on >>>> > EC2 without a problem. >>>> > >>>> > /b >>>> >>>> Hey Brian, >>>> >>>> FreeSWITCH in EC2 is a bit of a mystery to me... >>>> >>>> Call me old fashioned but in my mind VoIP and geography are linked >>>> in %99 of scenarios. Having VoIP services in a pure "cloud" >>>> environment just doesn't sound like a good idea to me. >>>> >>>> Consider a "hosted" environment with clients registered to a >>>> FreeSWITCH server. One day your instance is physically running on >>>> hardware in Seattle. The next day it could (potentially) be running >>>> in Chicago. That's obviously a very different routing path for your >>>> clients. Even /if/ Amazon (or whomever) employs every routing/network >>>> trick in the book you still won't be able to get over that change in >>>> geography. >>>> >>>> It's certainly possible a change like this may very well never >>>> happen in practice. I wouldn't know; I've never used EC2 and I don't >>>> even know that much about it. I'm just curious how well strictly, >>>> practically speaking this will work in the long term. >>>> >>>> -- >>>> Kristian Kielhofner >>>> http://www.astlinux.org >>>> http://blog.krisk.org >>>> http://www.star2star.com >>>> http://www.submityoursip.com >>>> http://www.voalte.com >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/41c395e8/attachment.html From drew.ozier at gmail.com Wed May 27 12:27:03 2009 From: drew.ozier at gmail.com (Drew Ozier) Date: Wed, 27 May 2009 15:27:03 -0400 Subject: [Freeswitch-users] Double-dtmf detection in IVR when a call is routed through FreeSWITCH In-Reply-To: <2388e50e0905270911u7b009f9fy4f7d7abfc7b781e9@mail.gmail.com> References: <2388e50e0905261336s120f568ek266ba4ecf9ffec01@mail.gmail.com> <2388e50e0905270802x337b6eb7w3e46ce2c6d491ab3@mail.gmail.com> <247344A9-0FE1-4964-8365-34874C194432@freeswitch.org> <2388e50e0905270911u7b009f9fy4f7d7abfc7b781e9@mail.gmail.com> Message-ID: <2388e50e0905271227r7f36b2d7w802e292fb35a0001@mail.gmail.com> Did a few TCP dumps, and before, I was seeing explicit DTMF packets coming out of my FreeSWITCH machine. Removing the only things I could find that had to do with 2833 in my internal and external sip_profiles (namely rfc2833-pt), I find that I don't see any mention of switch_rtp.c sending any DTMF packets, nor do I see any in the TCP dump. This still hasn't solved my problem. I tested again, and with 'start_dtmf', I get an audible hiccup in the audio when DTMF occurs. When I don't use 'start_dtmf', the hiccup goes away. Is there anything else that I should be turning off on FreeSWITCH to avoid this hiccup? -Drew On Wed, May 27, 2009 at 12:11 PM, Drew Ozier wrote: > The AudioCodes machines are always a bit of a nuisance to get acting the > way I want, but I think I have them sending all DTMF in-band. How do I > disable 2833 in my sofia profile? > > -Drew > > > On Wed, May 27, 2009 at 11:52 AM, Brian West wrote: > >> I highly recommend you DO NOT use inband AT ALL if possible. out of band >> is king. >> /b >> >> On May 27, 2009, at 10:49 AM, Milena wrote: >> >> Are the configurations of the audiocodes set to send only in-audio dtmf? >> make sure everything is sending only in-audio >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/3b638c76/attachment.html From anthony.minessale at gmail.com Wed May 27 12:38:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 27 May 2009 14:38:10 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <23746348.post@talk.nabble.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <11990ade0905261817t5e7c63fal745d8492604cbb73@mail.gmail.com> <191c3a030905270550s140021fbg21c3b065828e5995@mail.gmail.com> <11990ade0905270927o4d0a15edm750d46aef1c9bc0@mail.gmail.com> <23746348.post@talk.nabble.com> Message-ID: <191c3a030905271238u4c00e901te193818891a38034@mail.gmail.com> It would be an improvement to move the caller controls to the member so it works the way you expect but it will have to wait for the right time and motivation level. >From there it would also be possible to make up some new caller controls that were moderator inspired like (mute all besides moderator) stuff like that. I don't want to get too fancy because the idea behind the conference is to keep it clean. We do have a caller control that just sends an event so an external application can catch it and use esl to do the desired action such as, dial 4, get the event on the socket decide it means mute all and then send the correct series of api calls to mute each member. On Wed, May 27, 2009 at 11:45 AM, j3flight wrote: > > Nope, it seems you are absolutley correct. I had setup my conferences this > way, but hadn't experimented yet. I tried putting two users into the same > conference using different profiles, but they both had the same caller > controls. Bummer. > > I believe we can get around this though, assuming I can call javascript > from > the caller controls. Currently, I am using a trick in my conference IVR > where I scan the UUIDs in a conference looking for an IsModerator variable > that I set when they were dropped in. If a user has that variable set to > 1, > they are a moderator. > > I could use that same trick from the caller controls to determine whether > that user has moderator permissions and can act (or not act) on their > button > press. I'll give it a shot at some point and post back. > > > > > Stephen Crosby-2 wrote: > > > > Thanks j3flight, I have used that method, but the profiles seem to be for > > conferences not users. So I give a conference a profile, and everybody in > > the conference shares the profile settings. If I'm wrong, let me know, > > otherwise I'll stop hijacking this thread. > > > > --Stephen > > > > On Wed, May 27, 2009 at 5:50 AM, Anthony Minessale < > > anthony.minessale at gmail.com> wrote: > > > >> This opens the door to what you ask but to add moderator specific dtmf > >> controls is another patch that would require specifics and most likely a > >> bounty or contract to complete. > >> > >> > >> > >> On Tue, May 26, 2009 at 8:17 PM, Stephen Crosby > >> wrote: > >> > >>> Michael, > >>> > >>> Pardon me for hopping on this thread, but can you explain more about > >>> this > >>> new feature? I've been wanting something like this to apply different > >>> behaviors for different conference members. Can this be used to provide > >>> a > >>> 'moderator' with different behaviors bound to DTMF keys than regular > >>> callers? > >>> > >>> --Stephen > >>> > >>> On Tue, May 26, 2009 at 5:55 PM, Michael Collins > >>> wrote: > >>> > >>>> > >>>> > >>>> On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < > >>>> anthony.minessale at gmail.com> wrote: > >>>> > >>>>> the easiest way would be the new feature I added to 13442 > >>>>> > >>>>> in the conference profile add > >>>>> > >>>>> > >>>>> > >>>>> to your > >>>>> > >>>>> and in your dialplan > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> or > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> Don't forget the wishlist and donate button on the main site.... > >>>>> > >>>>> > >>>> And the wiki tax if you feel comfortable adding this to the wiki. If > >>>> not > >>>> let me know and I'll add it. > >>>> -MC > >>>> > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at 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 at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > >> iax:guest at conference.freeswitch.org/888 > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> pstn:213-799-1400 > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23746348.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/33e68fea/attachment-0001.html From larclap at yahoo.com Wed May 27 15:56:56 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 27 May 2009 15:56:56 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <87f2f3b90905271155t3bcdc087g986066949580dac7@mail.gmail.com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> <00c601c9def8$ee5eb690$cb1c23b0$@com> <87f2f3b90905271155t3bcdc087g986066949580dac7@mail.gmail.com> Message-ID: <010701c9df1e$6f8acc70$4ea06550$@com> When I create an xml file in conf/dialplan/default, I assume that it's 'inheriting' its context, right? If that's so, do I use this structure, for example conf/dialplan/default/00_MyExtension.xml: . Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 27, 2009 11:56 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Questions on build 13441 On Wed, May 27, 2009 at 11:28 AM, Lars Zeb wrote: Michael, thanks for fixing the links. And what about your advice on defining an extension that pertains all internal extensions (1000-1019) which I have currently put in conf/dialplan/default.xml? Is there an alternative file I should be putting this into? If you need to change the default local_extensions dp entry then there's not a whole lot of options. You could manually delete (or carefully comment out) all of the default stuff and then create your own custom version of that extension and place it in conf/dialplan/default/00_MyExtension.xml or something like that... The reason I say carefully comment out is that you cannot nest comments in XML. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/5bcaedde/attachment.html From neale at lowendale.com.au Wed May 27 15:51:50 2009 From: neale at lowendale.com.au (Neale Banks) Date: Thu, 28 May 2009 08:51:50 +1000 (EST) Subject: [Freeswitch-users] FS PABX experiences? Message-ID: Hi, We're considering deploying FS instead of a traditional PABX/Key-System in a small office environment (i.e. primarily non-technical users, 15-20 handsets). Anyone have any experiences (good/bad/whatever) in this sort of scenario? Thanks, Neale. From brian at freeswitch.org Wed May 27 16:06:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 18:06:57 -0500 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <010701c9df1e$6f8acc70$4ea06550$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> <00c601c9def8$ee5eb690$cb1c23b0$@com> <87f2f3b90905271155t3bcdc087g986066949580dac7@mail.gmail.com> <010701c9df1e$6f8acc70$4ea06550$@com> Message-ID: Order matters if something is loaded higher that matches which the debug log shows you this if you press F8 /b On May 27, 2009, at 5:56 PM, Lars Zeb wrote: > When I create an xml file in conf/dialplan/default, I assume that > it?s ?inheriting? its context, right? > > If that?s so, do I use this structure, for example conf/dialplan/ > default/00_MyExtension.xml: > > > > ? > > > > Thanks, Lars > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/ee44c5a6/attachment.html From msc at freeswitch.org Wed May 27 16:09:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 May 2009 16:09:30 -0700 Subject: [Freeswitch-users] Questions on build 13441 In-Reply-To: <010701c9df1e$6f8acc70$4ea06550$@com> References: <000301c9de5e$88c51800$9a4f4800$@com> <87f2f3b90905261757w3a463ebav60030a0a09b39d97@mail.gmail.com> <002701c9dede$11452560$33cf7020$@com> <98A71ADA-E933-42FC-B8E2-624FE67E43C1@freeswitch.org> <87f2f3b90905271004wa8a8e77v44c96b0187706fd5@mail.gmail.com> <00c601c9def8$ee5eb690$cb1c23b0$@com> <87f2f3b90905271155t3bcdc087g986066949580dac7@mail.gmail.com> <010701c9df1e$6f8acc70$4ea06550$@com> Message-ID: <87f2f3b90905271609n54694a1erc3cb238909c69ad9@mail.gmail.com> On Wed, May 27, 2009 at 3:56 PM, Lars Zeb wrote: > When I create an xml file in conf/dialplan/default, I assume that it?s > ?inheriting? its context, right? > > > Correct. Files in conf/dialplan/default/*xml are included in the default context while conf/dialplan/public/*xml are included in the public context. You may also create your own contexts if need be. > If that?s so, do I use this structure, for example > conf/dialplan/default/00_MyExtension.xml: > > > > > > > > ? > > > > > > > Yes, that is perfectly fine. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/d8175c36/attachment.html From msc at freeswitch.org Wed May 27 16:13:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 May 2009 16:13:14 -0700 Subject: [Freeswitch-users] FS PABX experiences? In-Reply-To: References: Message-ID: <87f2f3b90905271613h48777f00n579b8cb0a6121853@mail.gmail.com> On Wed, May 27, 2009 at 3:51 PM, Neale Banks wrote: > Hi, > > We're considering deploying FS instead of a traditional PABX/Key-System in > a small office environment (i.e. primarily non-technical users, 15-20 > handsets). > > Anyone have any experiences (good/bad/whatever) in this sort of scenario? > We have several people here who administer such systems. The key, though, is that someone will need to be the administrator of that system, especially since you have primarily non-technical users. Also, there isn't a FS GUI front-end yet (unless you count emacs) so you'll need to be prepared for a little text-based setup. All that being said, FreeSWITCH is a very stable, feature-rich platform. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/a3a57624/attachment-0001.html From jim at evolutiontel.net Wed May 27 16:22:56 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 28 May 2009 09:22:56 +1000 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: Message-ID: Hey Brian, I dug it! Regards, On Wed, May 27, 2009 at 5:47 AM, Brian West wrote: > Dear FreeSWITCHers, > Now I'm gonna take a moment here to guilt each and everyone of you into > checking out the story about Pre8 on Digg. ?We have all worked long and hard > to get to 1.0.4 and we still have a little bit to go. ?So everyone out there > that asks "What can I do to help the project?", this is your chance to do > so. ?Help us to promote the project, which brings more people to help in > supporting the community, the project and you the end user. > Also looking for people to help manage jira, test bugs, ask the right > questions and line up the bugs so we can knock them out! Please email me if > you're interested. > Here is the link for you to help > out?http://digg.com/search?s=FreeSWITCH+Pre8 > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed May 27 16:35:10 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 18:35:10 -0500 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: Message-ID: <96625A43-34FA-49AD-9D22-E25F78F6CA74@freeswitch.org> Thank you... now let me explain how this works. The more DIGG's you get in a short period of time... the more the chance you get on the front page. If you don't get that in the first 5 hours you're pretty much SOL. :P So there was a bit of urgency to that... maybe we'll be ready for the 1.0.4 release. /b On May 27, 2009, at 6:22 PM, Jim Burke wrote: > Hey Brian, > > I dug it! > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/6eec9328/attachment.html From jim at evolutiontel.net Wed May 27 16:52:02 2009 From: jim at evolutiontel.net (Jim Burke) Date: Thu, 28 May 2009 09:52:02 +1000 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: <96625A43-34FA-49AD-9D22-E25F78F6CA74@freeswitch.org> References: <96625A43-34FA-49AD-9D22-E25F78F6CA74@freeswitch.org> Message-ID: What can I say, first time digger :( On Thu, May 28, 2009 at 9:35 AM, Brian West wrote: > Thank you... now let me explain how this works. ?The more DIGG's you get in > a short period of time... the more the chance you get on the front page. > If you don't get that in the first 5 hours you're pretty much SOL. :P ?So > there was a bit of urgency to that... maybe we'll be ready for the 1.0.4 > release. > /b > On May 27, 2009, at 6:22 PM, Jim Burke wrote: > > Hey Brian, > > I dug it! > > Regards, > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gcd at i.ph Wed May 27 16:53:28 2009 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 28 May 2009 07:53:28 +0800 Subject: [Freeswitch-users] FS PABX experiences? In-Reply-To: References: Message-ID: <7d0bfd8c0905271653r72c83e03u24305a3c101be582@mail.gmail.com> IMHO, you have tons of features w/ FS. i've setup FS on a low-power consumption Intel D945GCLF2 motherboard (Atom dual-core CPU) ideal for 24/7 operation on a 10-seat contact center w/ default conversation recording. no problem. another cool feature. you can route the call based on the Caller ID. so u hv to consider the selection of the telco (FXO) gateway. one advantage over key system - you can turn PCs into extension phones using free softphones. just use USB phones instead of headsets. re maintenance, just provide remote access to the FS box. in my home FS, i create dialplan to reboot or shutdown my FS. it helps when problems occur (not encountered so far). -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Thu, May 28, 2009 at 6:51 AM, Neale Banks wrote: > Hi, > > We're considering deploying FS instead of a traditional PABX/Key-System in > a small office environment (i.e. primarily non-technical users, 15-20 > handsets). > > Anyone have any experiences (good/bad/whatever) in this sort of scenario? > > Thanks, > Neale. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/de595d37/attachment.html From daletrub at gmail.com Wed May 27 16:58:50 2009 From: daletrub at gmail.com (Dale Trub) Date: Wed, 27 May 2009 16:58:50 -0700 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> Message-ID: Anthony, Thank you for your suggestions! We are working on 1), but need to re-integrate code we've changed, and do regression testing. That's in progress, and we expect to be able to upgrade by the end of next week. We did manage to do 3) and 4), and we now have SIP logs (attached). Are you able to see anything that's out of the ordinary that we should be paying attention to? Best, Dale On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > 1) update to lastest trunk (you are at least 1000 revisions behind) > 2) disable the presence debug in sofia.conf > 3) enable sip trace instead "sofia profile internal siptrace on" > 4) reproduce your problem. > > Make sure you include more of the log from before the hangup happened. > The one you posted here is missing some of the info from the few seconds > prior but with the incomplete > info it looks like the other side sent a BYE ending the call. > > > > On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: > >> Thanks Brian! To answer your questions: >> Freeswitch svn revision: 12148 >> Centos rev: 2.6.18-92.el5 >> >> And apologies, actually I guess we're using g711 not 729. >> >> Jason: I agree it would seem to be on the switch/telco side. And, the >> telco says many other people are in the same set-up as us and don't have any >> issues, so they're insisting it's on our end. >> >> On Thu, May 21, 2009 at 7:28 PM, Brian West wrote: >> >>> >>> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >>> >>> We're running FreeSwitch as part of a teleconferencing service, inside a >>> telcom (so no >>> internet latency/NAT issues) and using g.729 >>> >>> >>> So you're using g729 with conferences? >>> >>> We are receiving some complaints of dropped calls, >>> including from landlines. This means they join the conference, and x >>> minutes in they simply drop. >>> >>> I know that cellphones tend to drop calls frequently, but landlines >>> are pretty reliable, and we're hearing it a lot. From the FreeSwitch >>> side of things, it just >>> looks like those callers hung up (but then dialed back in just a moment >>> later). >>> >>> I'm attaching two different snippets of the FS log files where these >>> issues are occurring. >>> >>> >>> Next time please call them .txt because you cause extra work to have to >>> open them otherwise. >>> >>> Does anyone have any recommendations about how to troubleshoot this? >>> >>> Any known issues/patches in FS that could be biting us? >>> >>> >>> Depends you failed to include some very valid info such as what version >>> or svn rev you're running and what linux distro. >>> >>> Is there some SIP logging we can do to debug? >>> >>> >>> Yes covered on the wiki. >>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>> >>> Are there any paid contractors avail who would have the expertise to look >>> into this? >>> >>> >>> email consulting at freeswitch.org >>> >>> Any help appreciated ... this is a major issue for us! >>> >>> Thanks much, >>> >>> -Dale >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/997e060f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sanitized-logs.tgz Type: application/x-gzip Size: 24555 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/997e060f/attachment-0001.tgz From jmesquita at gmail.com Wed May 27 16:59:06 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 27 May 2009 20:59:06 -0300 Subject: [Freeswitch-users] FS PABX experiences? In-Reply-To: <7d0bfd8c0905271653r72c83e03u24305a3c101be582@mail.gmail.com> References: <7d0bfd8c0905271653r72c83e03u24305a3c101be582@mail.gmail.com> Message-ID: <5a8712120905271659i35e3f455i9fddea8b7a408e0d@mail.gmail.com> I use it on a 12 extension office. Works like a charm. Specially because I host it on a cheap dedicated server (iWEB). The only thing I would say is to be careful not to loose focus on your primary business and start developing your own GUI for the pbx. I have seen that happen with lots of companies. They eventually fail. Mesquita On Wed, May 27, 2009 at 8:53 PM, Nandy Dagondon wrote: > IMHO, you have tons of features w/ FS. i've setup FS on a low-power > consumption Intel D945GCLF2 motherboard (Atom dual-core CPU) ideal for 24/7 > operation on a 10-seat contact center w/ default conversation recording. no > problem. > > another cool feature. you can route the call based on the Caller ID. so u > hv to consider the selection of the telco (FXO) gateway. > > one advantage over key system - you can turn PCs into extension phones > using free softphones. just use USB phones instead of headsets. > > re maintenance, just provide remote access to the FS box. in my home FS, i > create dialplan to reboot or shutdown my FS. it helps when problems occur > (not encountered so far). > > -nandy > =============================== > LanVox Systems > Lapulapu City, Philippines 6015 > Mobile: +63-920-6373450 > Phone: +63-32-3401807 > USA: +1-360-8122281 > http://sites.google.com/site/lanvoxphils > > > > > On Thu, May 28, 2009 at 6:51 AM, Neale Banks wrote: > >> Hi, >> >> We're considering deploying FS instead of a traditional PABX/Key-System in >> a small office environment (i.e. primarily non-technical users, 15-20 >> handsets). >> >> Anyone have any experiences (good/bad/whatever) in this sort of scenario? >> >> Thanks, >> Neale. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/c005bfb2/attachment.html From brian at freeswitch.org Wed May 27 17:04:33 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 27 May 2009 19:04:33 -0500 Subject: [Freeswitch-users] Pre8 Release on Digg In-Reply-To: References: <96625A43-34FA-49AD-9D22-E25F78F6CA74@freeswitch.org> Message-ID: Keep that mouse clicker ready! :) /b On May 27, 2009, at 6:52 PM, Jim Burke wrote: > What can I say, first time digger :( Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/7a2dab19/attachment.html From jmesquita at gmail.com Wed May 27 18:04:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 27 May 2009 22:04:50 -0300 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <23715721.post@talk.nabble.com> References: <23715721.post@talk.nabble.com> Message-ID: <5a8712120905271804s2e7aa3cci514cc490adad0ee7@mail.gmail.com> Quoting Mr. Anthony Minessale: thThe easiest way would be the new feature I added to 13442 > > in the conference profile add > > > > to your > > and in your dialplan > > > > > or > > > > > Don't forget the wishlist and donate button on the main site.... > On Tue, May 26, 2009 at 10:20 AM, j3flight wrote: > > I'm attempting to replicate the behavior of an Asterisk system with > FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to > be supported (easily). > > Ok, so I've setup my dialplan so that when a specific extension is hit, it > calls out to some javascript which acts like an IVR to handle the > conference > setup. (Similar to this: > http://wiki.freeswitch.org/wiki/Examples_confcall_js but with my own > improvements.) Anyway, the conferences are stored permanently in a > database, but I want them protected by their "owner" so they can only be > used when that conference owner dials. If other users have entered the > conference prior to the owner, they should hear music-on-hold until the > leader enters. > > This is easy in Asterisk because you can pop someone into MeetMe with > different flags. So, in my IVR, I prompt for the "conference number" > (known > to all) and then the "password" (known only to the owner/leader). If the > proper password is entered, the user is sent to conference XYZ with the > leader flag set. If no password is entered, the user goes to conference > XYZ, without the leader flag. If anyone enters before the leader, they're > told by MeetMe that "the conference will begin when the leader arrives" and > MeetMe provides MOH until that time. > > Help! This is an absolute deal-breaker for my install... How can I do > this > in FreeSwitch? > Thanks... > -- > View this message in context: > http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23715721p23715721.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090527/affaa737/attachment.html From jcromes at gmail.com Wed May 27 20:00:27 2009 From: jcromes at gmail.com (j3flight) Date: Wed, 27 May 2009 20:00:27 -0700 (PDT) Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> Message-ID: <23754379.post@talk.nabble.com> Wiki Tax paid... That was my first contribution to the freeswitch wiki! MC, you're welcome to have a look over it and see if i made things clear enough. Feel free to edit. On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale < And the wiki tax if you feel comfortable adding this to the wiki. If not let me know and I'll add it. -MC -- View this message in context: http://www.nabble.com/Conference-users-hear-MOH-until-leader-enters--tp23724561p23754379.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kiste at kiste.org Wed May 27 23:45:26 2009 From: kiste at kiste.org (Uwe Kastens) Date: Thu, 28 May 2009 08:45:26 +0200 Subject: [Freeswitch-users] fax@pstn with freeswitch Message-ID: <4A1E3306.3060800@kiste.org> Hello list, Since I tested the last year lot of T38 implementation with cpe, asterisk etc.pp. with lots of problems, I would like to have a solution to handle the fax issue with pstn. How is the mod_fax integrated with freeswitch? Is there a kind of "softmodem" needed, or could fax directly handled on a circuit of a sangoma E1 card? I would like to build a kind of routing device which could handle DDI correctly and is able to hold calls on TDM for fax. Example: I have a kind of numbers like 12345-[0-6] The user has a fax number on 12345-523 and 12345-455. Then I would like to handle 12345523 => fax-mod at tdm 12345455 => fax-mod at tdm 12345[0-6] => SIP BR Uwe -- kiste lat: 54.322684, lon: 10.13586 From saigop at gmail.com Thu May 28 00:03:19 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 28 May 2009 12:33:19 +0530 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <191c3a030905270601l5d451e61y1c09afd910cb86e5@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> <191c3a030905270601l5d451e61y1c09afd910cb86e5@mail.gmail.com> Message-ID: <2ea4d47e0905280003u5ac692f3h4956db2a1ead20af@mail.gmail.com> Let me explain in different way. As you suggested I tried the sessionX.setAutoHangup(0); and it works fine. I can able to transfer the call by uuid_transfer by which the call is originated with javascript via event socket. But the problem is once the call gets transferred, the extension has to hangup automatically rite? Thats not happening. On Wed, May 27, 2009 at 6:31 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I am not sure what you mean at this point. > > > > On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N wrote: > >> Hi Anthony, >> thanks, it seems to be working, but the extension is not hanging up once >> I transfered the call to another mobile or to a conference. Some where I am >> wrong? >> >> On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> This one happens to every new guy trying to make FS into a dialer app >>> using JS. >>> >>> for every sessionX you create in js with the new Session constructor >>> >>> sessionX.setAutoHangup(0); >>> >>> This allows the channels to remain alive outside the script once they are >>> hungup/transferred etc. >>> >>> >>> On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: >>> >>>> Hi, >>>> I had some discussion with the IRC regarding the uuid_transfer gets >>>> hang-up where the call is originated via javascript thru event socket. I was >>>> suggested to install latest SVN trunk. I did that again the same issue, the >>>> log is attached with here http://pastebin.freeswitch.org/9103 >>>> >>>> My call flow like this, >>>> >>>> 1. api jsrun fils.js >>>> 2. capture the uuid >>>> 3. api uuid_transfer -both >>>> >>>> Both the leg gets hangedup. >>>> >>>> Someone can assist me where I am wrong. >>>> >>>> -- >>>> Thank you with regards, >>>> Gopal, >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Thank you with regards, >> Gopal, >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/7f2c88b0/attachment-0001.html From anthony.minessale at gmail.com Thu May 28 06:07:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 May 2009 08:07:27 -0500 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> Message-ID: <191c3a030905280607q7dd0b5b7xc9e3f02b9f6c8824@mail.gmail.com> Also you should be putting these details in a jira report. http://jira.freeswitch.org open an issue report and attach all relevant logs, do not attach tarballs or gzipped files and make sure text files have a .txt extension. On Wed, May 27, 2009 at 6:58 PM, Dale Trub wrote: > Anthony, > > Thank you for your suggestions! We are working on 1), but need to > re-integrate code we've changed, and do regression testing. That's in > progress, and we expect to be able to upgrade by the end of next week. > > We did manage to do 3) and 4), and we now have SIP logs (attached). Are you > able to see anything that's out of the ordinary that we should be paying > attention to? > > Best, > Dale > > On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> 1) update to lastest trunk (you are at least 1000 revisions behind) >> 2) disable the presence debug in sofia.conf >> 3) enable sip trace instead "sofia profile internal siptrace on" >> 4) reproduce your problem. >> >> Make sure you include more of the log from before the hangup happened. >> The one you posted here is missing some of the info from the few seconds >> prior but with the incomplete >> info it looks like the other side sent a BYE ending the call. >> >> >> >> On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: >> >>> Thanks Brian! To answer your questions: >>> Freeswitch svn revision: 12148 >>> Centos rev: 2.6.18-92.el5 >>> >>> And apologies, actually I guess we're using g711 not 729. >>> >>> Jason: I agree it would seem to be on the switch/telco side. And, the >>> telco says many other people are in the same set-up as us and don't have any >>> issues, so they're insisting it's on our end. >>> >>> On Thu, May 21, 2009 at 7:28 PM, Brian West wrote: >>> >>>> >>>> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >>>> >>>> We're running FreeSwitch as part of a teleconferencing service, inside a >>>> telcom (so no >>>> internet latency/NAT issues) and using g.729 >>>> >>>> >>>> So you're using g729 with conferences? >>>> >>>> We are receiving some complaints of dropped calls, >>>> including from landlines. This means they join the conference, and x >>>> minutes in they simply drop. >>>> >>>> I know that cellphones tend to drop calls frequently, but landlines >>>> are pretty reliable, and we're hearing it a lot. From the FreeSwitch >>>> side of things, it just >>>> looks like those callers hung up (but then dialed back in just a moment >>>> later). >>>> >>>> I'm attaching two different snippets of the FS log files where these >>>> issues are occurring. >>>> >>>> >>>> Next time please call them .txt because you cause extra work to have to >>>> open them otherwise. >>>> >>>> Does anyone have any recommendations about how to troubleshoot this? >>>> >>>> Any known issues/patches in FS that could be biting us? >>>> >>>> >>>> Depends you failed to include some very valid info such as what version >>>> or svn rev you're running and what linux distro. >>>> >>>> Is there some SIP logging we can do to debug? >>>> >>>> >>>> Yes covered on the wiki. >>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>> >>>> Are there any paid contractors avail who would have the expertise to >>>> look into this? >>>> >>>> >>>> email consulting at freeswitch.org >>>> >>>> Any help appreciated ... this is a major issue for us! >>>> >>>> Thanks much, >>>> >>>> -Dale >>>> >>>> >>>> Brian West >>>> brian at freeswitch.org >>>> >>>> -- Meet us at ClueCon! http://www.cluecon.com >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/d203cddb/attachment.html From anthony.minessale at gmail.com Thu May 28 06:08:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 May 2009 08:08:23 -0500 Subject: [Freeswitch-users] calls appear to be dropping ... from landlines In-Reply-To: <191c3a030905280607q7dd0b5b7xc9e3f02b9f6c8824@mail.gmail.com> References: <12ED2D90-9D8F-442D-8A88-3754989185A6@freeswitch.org> <191c3a030905220547q10300638k6e55063c79b2ce8b@mail.gmail.com> <191c3a030905280607q7dd0b5b7xc9e3f02b9f6c8824@mail.gmail.com> Message-ID: <191c3a030905280608t50ccbf61gea8f36090ec4f6b1@mail.gmail.com> btw, 3 and 4 are not useful without 1 we only debug issues with svn trunk On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Also you should be putting these details in a jira report. > http://jira.freeswitch.org > > open an issue report and attach all relevant logs, do not attach tarballs > or gzipped files and make sure text files have a .txt extension. > > > > On Wed, May 27, 2009 at 6:58 PM, Dale Trub wrote: > >> Anthony, >> >> Thank you for your suggestions! We are working on 1), but need to >> re-integrate code we've changed, and do regression testing. That's in >> progress, and we expect to be able to upgrade by the end of next week. >> >> We did manage to do 3) and 4), and we now have SIP logs (attached). Are >> you able to see anything that's out of the ordinary that we should be paying >> attention to? >> >> Best, >> Dale >> >> On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> 1) update to lastest trunk (you are at least 1000 revisions behind) >>> 2) disable the presence debug in sofia.conf >>> 3) enable sip trace instead "sofia profile internal siptrace on" >>> 4) reproduce your problem. >>> >>> Make sure you include more of the log from before the hangup happened. >>> The one you posted here is missing some of the info from the few seconds >>> prior but with the incomplete >>> info it looks like the other side sent a BYE ending the call. >>> >>> >>> >>> On Thu, May 21, 2009 at 10:09 PM, Dale Trub wrote: >>> >>>> Thanks Brian! To answer your questions: >>>> Freeswitch svn revision: 12148 >>>> Centos rev: 2.6.18-92.el5 >>>> >>>> And apologies, actually I guess we're using g711 not 729. >>>> >>>> Jason: I agree it would seem to be on the switch/telco side. And, the >>>> telco says many other people are in the same set-up as us and don't have any >>>> issues, so they're insisting it's on our end. >>>> >>>> On Thu, May 21, 2009 at 7:28 PM, Brian West wrote: >>>> >>>>> >>>>> On May 21, 2009, at 9:15 PM, Dale Trub wrote: >>>>> >>>>> We're running FreeSwitch as part of a teleconferencing service, inside >>>>> a telcom (so no >>>>> internet latency/NAT issues) and using g.729 >>>>> >>>>> >>>>> So you're using g729 with conferences? >>>>> >>>>> We are receiving some complaints of dropped calls, >>>>> including from landlines. This means they join the conference, and x >>>>> minutes in they simply drop. >>>>> >>>>> I know that cellphones tend to drop calls frequently, but landlines >>>>> are pretty reliable, and we're hearing it a lot. From the FreeSwitch >>>>> side of things, it just >>>>> looks like those callers hung up (but then dialed back in just a moment >>>>> later). >>>>> >>>>> I'm attaching two different snippets of the FS log files where these >>>>> issues are occurring. >>>>> >>>>> >>>>> Next time please call them .txt because you cause extra work to have to >>>>> open them otherwise. >>>>> >>>>> Does anyone have any recommendations about how to troubleshoot this? >>>>> >>>>> Any known issues/patches in FS that could be biting us? >>>>> >>>>> >>>>> Depends you failed to include some very valid info such as what version >>>>> or svn rev you're running and what linux distro. >>>>> >>>>> Is there some SIP logging we can do to debug? >>>>> >>>>> >>>>> Yes covered on the wiki. >>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch >>>>> >>>>> Are there any paid contractors avail who would have the expertise to >>>>> look into this? >>>>> >>>>> >>>>> email consulting at freeswitch.org >>>>> >>>>> Any help appreciated ... this is a major issue for us! >>>>> >>>>> Thanks much, >>>>> >>>>> -Dale >>>>> >>>>> >>>>> Brian West >>>>> brian at freeswitch.org >>>>> >>>>> -- Meet us at ClueCon! http://www.cluecon.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/c8120fba/attachment-0001.html From Claudio.Cavalera at italtel.it Thu May 28 06:48:13 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 28 May 2009 15:48:13 +0200 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Virtualization has issues with timing in my experiance. I also experimetned timing issues with virtualization. Since everybody in the world speaks a lot good of virtualization I sometime think I'm the only one thinking wrong about it. I once read this http://www.vmware.com/pdf/vmware_timekeeping.pdf but I did not understood if it was a temporary problem to be solved in later linux kernels or something here to stay. Ciao, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. From saigop at gmail.com Thu May 28 07:14:22 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 28 May 2009 19:44:22 +0530 Subject: [Freeswitch-users] Freeswitch with APR Message-ID: <2ea4d47e0905280714t4b539e69w70feb7d8642d2972@mail.gmail.com> Hi, I saw the apache portable runtime is included in freeswitch. So far I understand that using APR will give good performance. Am I correct? or it has been used for some other scenarios like even socket or dialplan? -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/40d4d326/attachment.html From anthony.minessale at gmail.com Thu May 28 08:21:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 May 2009 10:21:30 -0500 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <2ea4d47e0905280003u5ac692f3h4956db2a1ead20af@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> <191c3a030905270601l5d451e61y1c09afd910cb86e5@mail.gmail.com> <2ea4d47e0905280003u5ac692f3h4956db2a1ead20af@mail.gmail.com> Message-ID: <191c3a030905280821j1807e251ha6e5090f8fa9eeab@mail.gmail.com> That was pretty much a repeat of the same explanation. I am still not sure what you mean? What is the "call" and what is the "extension" and what is not hanging up? On Thu, May 28, 2009 at 2:03 AM, Gopalakrishnan A.N wrote: > Let me explain in different way. > As you suggested I tried the sessionX.setAutoHangup(0); and it works > fine. I can able to transfer the call by uuid_transfer by which the call is > originated with javascript via event socket. > But the problem is once the call gets transferred, the extension has to > hangup automatically rite? Thats not happening. > > > > On Wed, May 27, 2009 at 6:31 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I am not sure what you mean at this point. >> >> >> >> On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N wrote: >> >>> Hi Anthony, >>> thanks, it seems to be working, but the extension is not hanging up >>> once I transfered the call to another mobile or to a conference. Some where >>> I am wrong? >>> >>> On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> This one happens to every new guy trying to make FS into a dialer app >>>> using JS. >>>> >>>> for every sessionX you create in js with the new Session constructor >>>> >>>> sessionX.setAutoHangup(0); >>>> >>>> This allows the channels to remain alive outside the script once they >>>> are hungup/transferred etc. >>>> >>>> >>>> On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: >>>> >>>>> Hi, >>>>> I had some discussion with the IRC regarding the uuid_transfer gets >>>>> hang-up where the call is originated via javascript thru event socket. I was >>>>> suggested to install latest SVN trunk. I did that again the same issue, the >>>>> log is attached with here http://pastebin.freeswitch.org/9103 >>>>> >>>>> My call flow like this, >>>>> >>>>> 1. api jsrun fils.js >>>>> 2. capture the uuid >>>>> 3. api uuid_transfer -both >>>>> >>>>> Both the leg gets hangedup. >>>>> >>>>> Someone can assist me where I am wrong. >>>>> >>>>> -- >>>>> Thank you with regards, >>>>> Gopal, >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Thank you with regards, >>> Gopal, >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/9a253fdb/attachment.html From msc at freeswitch.org Thu May 28 08:56:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 28 May 2009 08:56:19 -0700 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <23754379.post@talk.nabble.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> Message-ID: <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> On Wed, May 27, 2009 at 8:00 PM, j3flight wrote: > > Wiki Tax paid... > That was my first contribution to the freeswitch wiki! > MC, you're welcome to have a look over it and see if i made things clear > enough. Feel free to edit. > Nicely done! Thanks for taking the time to create a wiki user and jump in with both feet. BTW, as you gain more practical experience with this wait-mod/moderator feature please feel free to come back and add any useful tidbits to the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/844214e3/attachment.html From mike at jerris.com Thu May 28 09:25:28 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 28 May 2009 12:25:28 -0400 Subject: [Freeswitch-users] Freeswitch with APR In-Reply-To: <2ea4d47e0905280714t4b539e69w70feb7d8642d2972@mail.gmail.com> References: <2ea4d47e0905280714t4b539e69w70feb7d8642d2972@mail.gmail.com> Message-ID: apr is used primarily for portability and utility, not performance. It is used heavily throughout the entire codebase in the core and all modules (via our abstraction layer). On May 28, 2009, at 10:14 AM, Gopalakrishnan A.N wrote: > Hi, > > I saw the apache portable runtime is included in freeswitch. So > far I understand that using APR will give good performance. Am I > correct? or it has been used for some other scenarios like even > socket or dialplan? From mgg at giagnocavo.net Thu May 28 09:49:17 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 28 May 2009 12:49:17 -0400 Subject: [Freeswitch-users] FS in Amazon EC2 for production? In-Reply-To: References: <7482D043-8C21-489A-AE84-16D4BF9C682B@gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262BF8822@mse17be1.mse17.exchange.ms> Yea, is anyone using VMware or Xen type deployments with heavy media or transcoding? -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cavalera Claudio Luigi Sent: Thursday, May 28, 2009 7:48 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in Amazon EC2 for production? freeswitch-users-bounces at lists.freeswitch.org wrote: > Virtualization has issues with timing in my experiance. I also experimetned timing issues with virtualization. Since everybody in the world speaks a lot good of virtualization I sometime think I'm the only one thinking wrong about it. I once read this http://www.vmware.com/pdf/vmware_timekeeping.pdf but I did not understood if it was a temporary problem to be solved in later linux kernels or something here to stay. Ciao, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gerry at pstn2.net Thu May 28 14:20:46 2009 From: gerry at pstn2.net (Gerry Hull) Date: Thu, 28 May 2009 17:20:46 -0400 Subject: [Freeswitch-users] Missing Events in mod_event_socket Message-ID: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> Hello, I am a Windows developer who has written an application around the event_socket interface. My client piece started with the C# EventSocket client Jonas Gauffin had posted on CodePlex. Well, Jonas did not keep up that code on Codeplex, but after communicating with him, I did get the latest client-side code from the freeaswitch SVN, and it seems to work fine. However, their is a persistent, nasty bug I'm seeing: On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which gives me some of the info I need on the incoming call. Following that event, I should get an EventChannelExecuteComplete event, which gives me important information like call-direction, channel-state, answer-state, caller-destination-number, caller-caller-id-name, etc. The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an inbound call, but EventChannelExecuteComplete does not fire --randomly. I thought this mighrt have something to do with linger, but executing the linger command does not help. Jonas made the following comment on the issue: "It has been a bug in the eventsocket implementation in freeswitch. It can sometimes skip packets if the socket layer in the os gives an error code (internal socket buffer becomes full). A simple send retry usually fixes the problem. I've created a patch for it long time ago (and reported it in FS jira). Mike Jerris have made an own fix for the issue. I do not know if it works, I'm still running my own patch. I've attached it to this email. It's a patch for freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, everything works gr8 for me with it." Well, I have no idea how to apply the patch. I've downloaded the latest code from trunk at files.freeswitch.org, and built FS using Visual Studio 2008. all compiles fine. However, the bug sticks it's nasty head up randomly about every other call. I've never done a patch... I tried downloading GNU Patch for windows, and tried applying it, but it reported errors. Has this issue been fixed in core code? If not, can someone help me patch this? I'm dead in the water on a project until I resolve this. In every other aspect, I've found FS to be flawless. Regards, Gerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/6d4c74a6/attachment.html From brian at freeswitch.org Thu May 28 14:36:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 28 May 2009 16:36:06 -0500 Subject: [Freeswitch-users] Missing Events in mod_event_socket In-Reply-To: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> References: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> Message-ID: Please report bugs to http://jira.freeswitch.org /b On May 28, 2009, at 4:20 PM, Gerry Hull wrote: > Hello, > > I am a Windows developer who has written an application around the > event_socket interface. My client piece started with the C# > EventSocket client Jonas Gauffin had posted on CodePlex. > Well, Jonas did not keep up that code on Codeplex, but after > communicating with him, I did get the latest client-side code from > the freeaswitch SVN, and it seems to work fine. > > However, their is a persistent, nasty bug I'm seeing: > > On an inbound call to FreeSwitch, I get the EventChannelAnswer > event, which gives me some of the info I need on the incoming call. > Following that event, I should get an EventChannelExecuteComplete > event, which gives me important information like call-direction, > channel-state, answer-state, caller-destination-number, caller- > caller-id-name, etc. > > The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an > inbound call, but EventChannelExecuteComplete does not fire -- > randomly. I thought this mighrt have something to do with linger, > but executing the linger command does not help. > > Jonas made the following comment on the issue: > > "It has been a bug in the eventsocket implementation in > freeswitch. It can sometimes skip packets if the socket layer in > the os gives an error code (internal socket buffer becomes full). > A simple send retry usually fixes the problem. I've created a patch > for it long time ago (and reported it in FS jira). Mike Jerris have > made an own fix for the issue. I do not know if it works, I'm still > running my own patch. I've attached it to this email. It's a patch > for freeswitch\src\mod\event_handlers\mod_event_socket\ > mod_event_socket.c, everything works gr8 for me with it." > > Well, I have no idea how to apply the patch. > > I've downloaded the latest code from trunk at files.freeswitch.org, > and built FS using Visual Studio 2008. all compiles fine. > However, the bug sticks it's nasty head up randomly about every > other call. > > I've never done a patch... I tried downloading GNU Patch for > windows, and tried applying it, but it reported errors. > > Has this issue been fixed in core code? If not, can someone help me > patch this? I'm dead in the water on a project until I resolve > this. In every other aspect, I've found FS to be flawless. > > Regards, > > Gerry Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/b52fa76f/attachment.html From anthony.minessale at gmail.com Thu May 28 16:05:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 May 2009 18:05:00 -0500 Subject: [Freeswitch-users] Missing Events in mod_event_socket In-Reply-To: References: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> Message-ID: <191c3a030905281605x45b0dad8g2840340f3482cb0@mail.gmail.com> and attach the patch in question On Thu, May 28, 2009 at 4:36 PM, Brian West wrote: > Please report bugs to http://jira.freeswitch.org > /b > > On May 28, 2009, at 4:20 PM, Gerry Hull wrote: > > Hello, > > I am a Windows developer who has written an application around the > event_socket interface. My client piece started with the C# EventSocket > client Jonas Gauffin had posted on CodePlex. > Well, Jonas did not keep up that code on Codeplex, but after communicating > with him, I did get the latest client-side code from the freeaswitch SVN, > and it seems to work fine. > > However, their is a persistent, nasty bug I'm seeing: > > On an inbound call to FreeSwitch, I get the EventChannelAnswer event, which > gives me some of the info I need on the incoming call. > Following that event, I should get an EventChannelExecuteComplete event, > which gives me important information like call-direction, > channel-state, answer-state, caller-destination-number, > caller-caller-id-name, etc. > > The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an > inbound call, but EventChannelExecuteComplete does not fire --randomly. I > thought this mighrt have something to do with linger, > but executing the linger command does not help. > > Jonas made the following comment on the issue: > > "It has been a bug in the eventsocket implementation in freeswitch. It > can sometimes skip packets if the socket layer in the os gives an error > code (internal socket buffer becomes full). > A simple send retry usually fixes the problem. I've created a patch for it > long time ago (and reported it in FS jira). Mike Jerris have made an own fix > for the issue. I do not know if it works, I'm still > running my own patch. I've attached it to this email. It's a patch for > freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, > everything works gr8 for me with it." > > Well, I have no idea how to apply the patch. > > I've downloaded the latest code from trunk at files.freeswitch.org, and > built FS using Visual Studio 2008. all compiles fine. However, the bug > sticks it's nasty head up randomly about every other call. > > I've never done a patch... I tried downloading GNU Patch for windows, and > tried applying it, but it reported errors. > > Has this issue been fixed in core code? If not, can someone help me patch > this? I'm dead in the water on a project until I resolve this. In every > other aspect, I've found FS to be flawless. > > Regards, > > Gerry > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/3825bef9/attachment-0001.html From anthony.minessale at gmail.com Thu May 28 16:33:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 28 May 2009 18:33:44 -0500 Subject: [Freeswitch-users] Missing Events in mod_event_socket In-Reply-To: <191c3a030905281605x45b0dad8g2840340f3482cb0@mail.gmail.com> References: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> <191c3a030905281605x45b0dad8g2840340f3482cb0@mail.gmail.com> Message-ID: <191c3a030905281633s236c2ad4if9902ea97365d484@mail.gmail.com> I dug up patch and it' clearly not the right patch and is only a self serving kludge for jonas. There is nothing wrong with that except he never tested our proper patch that only has on possible problem: the timeout being too short. I have updated the timeout to a much higher value please retest revision r13496 or greater On Thu, May 28, 2009 at 6:05 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > and attach the patch in question > > > On Thu, May 28, 2009 at 4:36 PM, Brian West wrote: > >> Please report bugs to http://jira.freeswitch.org >> /b >> >> On May 28, 2009, at 4:20 PM, Gerry Hull wrote: >> >> Hello, >> >> I am a Windows developer who has written an application around the >> event_socket interface. My client piece started with the C# EventSocket >> client Jonas Gauffin had posted on CodePlex. >> Well, Jonas did not keep up that code on Codeplex, but after communicating >> with him, I did get the latest client-side code from the freeaswitch SVN, >> and it seems to work fine. >> >> However, their is a persistent, nasty bug I'm seeing: >> >> On an inbound call to FreeSwitch, I get the EventChannelAnswer event, >> which gives me some of the info I need on the incoming call. >> Following that event, I should get an EventChannelExecuteComplete event, >> which gives me important information like call-direction, >> channel-state, answer-state, caller-destination-number, >> caller-caller-id-name, etc. >> >> The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an >> inbound call, but EventChannelExecuteComplete does not fire --randomly. I >> thought this mighrt have something to do with linger, >> but executing the linger command does not help. >> >> Jonas made the following comment on the issue: >> >> "It has been a bug in the eventsocket implementation in freeswitch. It >> can sometimes skip packets if the socket layer in the os gives an error >> code (internal socket buffer becomes full). >> A simple send retry usually fixes the problem. I've created a patch for it >> long time ago (and reported it in FS jira). Mike Jerris have made an own fix >> for the issue. I do not know if it works, I'm still >> running my own patch. I've attached it to this email. It's a patch for >> freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, >> everything works gr8 for me with it." >> >> Well, I have no idea how to apply the patch. >> >> I've downloaded the latest code from trunk at files.freeswitch.org, and >> built FS using Visual Studio 2008. all compiles fine. However, the bug >> sticks it's nasty head up randomly about every other call. >> >> I've never done a patch... I tried downloading GNU Patch for windows, and >> tried applying it, but it reported errors. >> >> Has this issue been fixed in core code? If not, can someone help me patch >> this? I'm dead in the water on a project until I resolve this. In every >> other aspect, I've found FS to be flawless. >> >> Regards, >> >> Gerry >> >> >> Brian West >> brian at freeswitch.org >> >> -- Meet us at ClueCon! http://www.cluecon.com >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/963e838b/attachment.html From gerry at pstn2.net Thu May 28 17:40:50 2009 From: gerry at pstn2.net (Gerry Hull) Date: Thu, 28 May 2009 20:40:50 -0400 Subject: [Freeswitch-users] Missing Events in mod_event_socket In-Reply-To: <191c3a030905281633s236c2ad4if9902ea97365d484@mail.gmail.com> References: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> <191c3a030905281605x45b0dad8g2840340f3482cb0@mail.gmail.com> <191c3a030905281633s236c2ad4if9902ea97365d484@mail.gmail.com> Message-ID: <98a86adf0905281740m10ee418cucc5ed30ec47ee2ff@mail.gmail.com> Thanks much Anthony, I'll do just that and report back. Gerry On Thu, May 28, 2009 at 7:33 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I dug up patch and it' clearly not the right patch and is only a self > serving kludge for jonas. > There is nothing wrong with that except he never tested our proper patch > that only has on possible problem: the timeout being too short. > > I have updated the timeout to a much higher value > > please retest revision r13496 or greater > > > > > > On Thu, May 28, 2009 at 6:05 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> and attach the patch in question >> >> >> On Thu, May 28, 2009 at 4:36 PM, Brian West wrote: >> >>> Please report bugs to http://jira.freeswitch.org >>> /b >>> >>> On May 28, 2009, at 4:20 PM, Gerry Hull wrote: >>> >>> Hello, >>> >>> I am a Windows developer who has written an application around the >>> event_socket interface. My client piece started with the C# EventSocket >>> client Jonas Gauffin had posted on CodePlex. >>> Well, Jonas did not keep up that code on Codeplex, but after >>> communicating with him, I did get the latest client-side code from the >>> freeaswitch SVN, and it seems to work fine. >>> >>> However, their is a persistent, nasty bug I'm seeing: >>> >>> On an inbound call to FreeSwitch, I get the EventChannelAnswer event, >>> which gives me some of the info I need on the incoming call. >>> Following that event, I should get an EventChannelExecuteComplete event, >>> which gives me important information like call-direction, >>> channel-state, answer-state, caller-destination-number, >>> caller-caller-id-name, etc. >>> >>> The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an >>> inbound call, but EventChannelExecuteComplete does not fire --randomly. I >>> thought this mighrt have something to do with linger, >>> but executing the linger command does not help. >>> >>> Jonas made the following comment on the issue: >>> >>> "It has been a bug in the eventsocket implementation in freeswitch. It >>> can sometimes skip packets if the socket layer in the os gives an error >>> code (internal socket buffer becomes full). >>> A simple send retry usually fixes the problem. I've created a patch for >>> it long time ago (and reported it in FS jira). Mike Jerris have made an own >>> fix for the issue. I do not know if it works, I'm still >>> running my own patch. I've attached it to this email. It's a patch for >>> freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, >>> everything works gr8 for me with it." >>> >>> Well, I have no idea how to apply the patch. >>> >>> I've downloaded the latest code from trunk at files.freeswitch.org, and >>> built FS using Visual Studio 2008. all compiles fine. However, the bug >>> sticks it's nasty head up randomly about every other call. >>> >>> I've never done a patch... I tried downloading GNU Patch for windows, and >>> tried applying it, but it reported errors. >>> >>> Has this issue been fixed in core code? If not, can someone help me >>> patch this? I'm dead in the water on a project until I resolve this. In >>> every other aspect, I've found FS to be flawless. >>> >>> Regards, >>> >>> Gerry >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/7eb54fe7/attachment-0001.html From dftoro at yahoo.com Thu May 28 17:49:00 2009 From: dftoro at yahoo.com (Diego Toro) Date: Thu, 28 May 2009 17:49:00 -0700 (PDT) Subject: [Freeswitch-users] The calls are dropped during register Message-ID: <335675.51213.qm@web33501.mail.mud.yahoo.com> Greetings ? The calls are dropped when FS try to register whith a pbx, I see mutiples requests of register from FS to pbx, although the register isn't lost, the calls are dropped. ? I have console trace: http://pastebin.freeswitch.org/9162 ? I appreciate any help ? Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/a56856ee/attachment.html From brian at freeswitch.org Thu May 28 17:54:49 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 28 May 2009 19:54:49 -0500 Subject: [Freeswitch-users] The calls are dropped during register In-Reply-To: <335675.51213.qm@web33501.mail.mud.yahoo.com> References: <335675.51213.qm@web33501.mail.mud.yahoo.com> Message-ID: <062EC6ED-88D2-4F5E-90C7-6B6D5BAA8138@freeswitch.org> Don't set your interval lower than 120 like you said.. if it hurts don't pick at it. /b On May 28, 2009, at 7:49 PM, Diego Toro wrote: > Greetings > > The calls are dropped when FS try to register whith a pbx, I see > mutiples requests of register from FS to pbx, although the register > isn't lost, the calls are dropped. > > I have console trace: http://pastebin.freeswitch.org/9162 > > I appreciate any help > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/2d788ab6/attachment.html From dftoro at yahoo.com Thu May 28 18:02:05 2009 From: dftoro at yahoo.com (Diego Toro) Date: Thu, 28 May 2009 18:02:05 -0700 (PDT) Subject: [Freeswitch-users] The calls are dropped during register Message-ID: <18122.60427.qm@web33501.mail.mud.yahoo.com> ? Setting?expire-seconds greater than or equal to 120 the register is lost. thank you ? Diego --- On Thu, 5/28/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] The calls are dropped during register To: freeswitch-users at lists.freeswitch.org Date: Thursday, May 28, 2009, 7:54 PM Don't set your interval lower than 120 like you said.. if it hurts don't pick at it. /b On May 28, 2009, at 7:49 PM, Diego Toro wrote: Greetings ? The calls are dropped when FS try to register whith a pbx, I see mutiples requests of register from FS to pbx, although the register isn't lost, the calls are dropped. ? I have console trace:?http://pastebin.freeswitch.org/9162 ? I appreciate any help ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/d49b9dff/attachment.html From brian at freeswitch.org Thu May 28 18:04:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 28 May 2009 20:04:42 -0500 Subject: [Freeswitch-users] The calls are dropped during register In-Reply-To: <18122.60427.qm@web33501.mail.mud.yahoo.com> References: <18122.60427.qm@web33501.mail.mud.yahoo.com> Message-ID: <737E3CA8-15B0-4545-B45B-54F7B6FF0B2F@freeswitch.org> Please add all these notes to the jira. I don't mind helping but its not on the top priority list right now as its only effecting two people with odd hardware. I need to know if others are seeing this behavior. /b On May 28, 2009, at 8:02 PM, Diego Toro wrote: > > Setting expire-seconds greater than or equal to 120 the register is > lost. > thank you > > Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/b62ef841/attachment.html From dftoro at yahoo.com Thu May 28 18:19:08 2009 From: dftoro at yahoo.com (Diego Toro) Date: Thu, 28 May 2009 18:19:08 -0700 (PDT) Subject: [Freeswitch-users] The calls are dropped during register Message-ID: <47990.71202.qm@web33504.mail.mud.yahoo.com> thank you Brian, ? my notes on http://jira.freeswitch.org/browse/SFSIP-143, I have hardware avaible for testing Diego --- On Thu, 5/28/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] The calls are dropped during register To: freeswitch-users at lists.freeswitch.org Date: Thursday, May 28, 2009, 8:04 PM Please add all these notes to the jira. ?I don't mind helping but its not on the top priority list right now as its only effecting two people with odd hardware. ?I need to know if others are seeing this behavior. /b On May 28, 2009, at 8:02 PM, Diego Toro wrote: ? Setting?expire-seconds greater than or equal to 120 the register is lost. thank you ? Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/68014fb9/attachment-0001.html From brian at freeswitch.org Thu May 28 18:22:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 28 May 2009 20:22:46 -0500 Subject: [Freeswitch-users] The calls are dropped during register In-Reply-To: <47990.71202.qm@web33504.mail.mud.yahoo.com> References: <47990.71202.qm@web33504.mail.mud.yahoo.com> Message-ID: <54F29E24-9155-475C-B885-C30ADAD73929@freeswitch.org> Anything on linux? /b On May 28, 2009, at 8:19 PM, Diego Toro wrote: > thank you Brian, > > my notes on http://jira.freeswitch.org/browse/SFSIP-143, I have > hardware avaible for testing > Diego > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/df2dee1a/attachment.html From gcd at i.ph Thu May 28 18:58:05 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 29 May 2009 09:58:05 +0800 Subject: [Freeswitch-users] Default IVR action Message-ID: <7d0bfd8c0905281858o7fd28d04q971972ff789b7945@mail.gmail.com> hi to all, i'm looking for a default action in the IVR in case the caller doesn't press any key. is this option available? with this option, we can add this prompt "... please stay on the line to be connected.". i know this can be done using scripts but it's better to have this feature on the app itself. rgds, -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/15ca18e5/attachment.html From brian at freeswitch.org Thu May 28 19:05:20 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 28 May 2009 21:05:20 -0500 Subject: [Freeswitch-users] Default IVR action In-Reply-To: <7d0bfd8c0905281858o7fd28d04q971972ff789b7945@mail.gmail.com> References: <7d0bfd8c0905281858o7fd28d04q971972ff789b7945@mail.gmail.com> Message-ID: Set the app after ivr to transfer and set the exit sound to "please stay on the line to be connected" /b On May 28, 2009, at 8:58 PM, Nandy Dagondon wrote: > hi to all, > > i'm looking for a default action in the IVR in case the caller > doesn't press any key. is this option available? with this option, > we can add this prompt "... please stay on the line to be > connected.". i know this can be done using scripts but it's better > to have this feature on the app itself. > > rgds, > -nandy Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/ad864edc/attachment.html From gcd at i.ph Thu May 28 19:39:50 2009 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 29 May 2009 10:39:50 +0800 Subject: [Freeswitch-users] Default IVR action In-Reply-To: References: <7d0bfd8c0905281858o7fd28d04q971972ff789b7945@mail.gmail.com> Message-ID: <7d0bfd8c0905281939w579406b0u7f916499020084a2@mail.gmail.com> tks brian. it worked. i changed the max-timeouts=1 for 1-pass announcement. -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils On Fri, May 29, 2009 at 10:05 AM, Brian West wrote: > Set the app after ivr to transfer and set the exit sound to "please stay on > the line to be connected"/b > > On May 28, 2009, at 8:58 PM, Nandy Dagondon wrote: > > hi to all, > > i'm looking for a default action in the IVR in case the caller doesn't > press any key. is this option available? with this option, we can add this > prompt "... please stay on the line to be connected.". i know this can be > done using scripts but it's better to have this feature on the app itself. > > > rgds, > -nandy > > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/fd166b3a/attachment.html From marc at kasteris.com Thu May 28 20:37:06 2009 From: marc at kasteris.com (Marc Orenberg) Date: Thu, 28 May 2009 20:37:06 -0700 (PDT) Subject: [Freeswitch-users] Connecting as a "POTS codec" to Prima LT Message-ID: <940350.55045.qm@web50812.mail.re2.yahoo.com> Hi, I'd like FreeSWITCH to be able?to communicate with a Musicam "Prima LT" device.? (http://www.musicamusa.com/products/prima/PrimaLT.htm). This is a "POTS codec", which (I've just learned) means that the connection is made via regular POTS phone connection, but instead of transmitting voice it transmits data packets, in this case G.722 data. Does anybody know if this is currently possible with FreeSWITCH, and if so, could someone point me in the right direction for establishing such a connection? Thanks in advance! Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090528/f8c5c061/attachment.html From jim at evolutiontel.net Thu May 28 21:40:58 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 29 May 2009 14:40:58 +1000 Subject: [Freeswitch-users] Contact Header Message-ID: Hi Gents, I have just upgraded to the latest trunk from 1.0.4Pre7 and found the Contact header in the 200Ok messages has changed from sip:mod_sofia at 123.123.123.123:5060 to dialed_number at sip.evolutiontel.net. Have looked at Jira and in old emails from the list and could not find anything, was there a variable that needs to added to sofia config to make sure the Contact header has an IP address instead of a URI? Thanks, Jim From mcampbellsmith at gmail.com Thu May 28 21:58:31 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 29 May 2009 14:58:31 +1000 Subject: [Freeswitch-users] gtalk text chat Message-ID: <33c87fa30905282158p43172ecfx40011f078519c44a@mail.gmail.com> Hi! Is it possible to configure freeswitch and mod_dingaling so that it sends a text chat to a gtalk client? I would plan to use this for certain debugging purposes. Thanks! /Mark From brian at freeswitch.org Thu May 28 22:25:17 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 00:25:17 -0500 Subject: [Freeswitch-users] Contact Header In-Reply-To: References: Message-ID: This behavior only takes place on , Which Is required in some cases for things like call pickup and barge when using TLS/TCP. Can you tell me why the contact with the hostname is giving you a problem? /b On May 28, 2009, at 11:40 PM, Jim Burke wrote: > Have looked at Jira and in old emails from the list and could not find > anything, was there a variable that needs to added to sofia config to > make sure the Contact header has an IP address instead of a URI? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/6b7567d3/attachment.html From brian at freeswitch.org Thu May 28 22:27:10 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 00:27:10 -0500 Subject: [Freeswitch-users] gtalk text chat In-Reply-To: <33c87fa30905282158p43172ecfx40011f078519c44a@mail.gmail.com> References: <33c87fa30905282158p43172ecfx40011f078519c44a@mail.gmail.com> Message-ID: http://www.freeswitch.org/eg/js/chat_js.txt Try something like that. /b On May 28, 2009, at 11:58 PM, Mark Campbell-Smith wrote: > Hi! > > Is it possible to configure freeswitch and mod_dingaling so that it > sends a text chat to a gtalk client? > > I would plan to use this for certain debugging purposes. > > Thanks! > /Mark Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/9e051f7b/attachment.html From brian at freeswitch.org Thu May 28 22:28:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 00:28:47 -0500 Subject: [Freeswitch-users] Connecting as a "POTS codec" to Prima LT In-Reply-To: <940350.55045.qm@web50812.mail.re2.yahoo.com> References: <940350.55045.qm@web50812.mail.re2.yahoo.com> Message-ID: <6C9C597F-D979-494F-9081-1D8DB95BE593@freeswitch.org> On May 28, 2009, at 10:37 PM, Marc Orenberg wrote: > Hi, I'd like FreeSWITCH to be able to communicate with a Musicam > "Prima LT" device. (http://www.musicamusa.com/products/prima/PrimaLT.htm > ). This is a "POTS codec", which (I've just learned) means that the > connection is made via regular POTS phone connection, but instead of > transmitting voice it transmits data packets, in this case G.722 data. Is this just a data pipe? Or what? > > Does anybody know if this is currently possible with FreeSWITCH, and > if so, could someone point me in the right direction for > establishing such a connection? > Yes FreeSWICH can do G.722 but not sure with that gear. > > Thanks in advance! > > Marc Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/5de15ea0/attachment.html From jim at evolutiontel.net Thu May 28 23:37:58 2009 From: jim at evolutiontel.net (Jim Burke) Date: Fri, 29 May 2009 16:37:58 +1000 Subject: [Freeswitch-users] Contact Header In-Reply-To: References: Message-ID: Thanks Brian, will check it out. I am using FS as Voicemail behind Opensips. As we have 2 Opensips servers if FS responds with a Contact header with a URI value we cannot route the call back to the correct FS server and the call is eventually dropped. For some reason this occurs even though we have record_route set in Opensips. Regards, On Fri, May 29, 2009 at 3:25 PM, Brian West wrote: > This behavior only takes place on? value="true"/>, Which Is required in some cases for things like call pickup > and barge when using TLS/TCP. ?Can you tell me why the contact with the > hostname is giving you a problem? > /b > On May 28, 2009, at 11:40 PM, Jim Burke wrote: > > Have looked at Jira and in old emails from the list and could not find > anything, was there a variable that needs to added to sofia config to > make sure the Contact header has an IP address instead of a URI? > > Brian West > brian at freeswitch.org > -- Meet us at ClueCon! ?http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jmesquita at gmail.com Fri May 29 00:18:46 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 29 May 2009 04:18:46 -0300 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> Message-ID: <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> I could not get this working on current trunk. Can you post your configuration on conference module and the dialplan example? Thanks, jmesquita On Thu, May 28, 2009 at 12:56 PM, Michael Collins wrote: > > > On Wed, May 27, 2009 at 8:00 PM, j3flight wrote: > >> >> Wiki Tax paid... >> That was my first contribution to the freeswitch wiki! >> MC, you're welcome to have a look over it and see if i made things clear >> enough. Feel free to edit. >> > > Nicely done! Thanks for taking the time to create a wiki user and jump in > with both feet. BTW, as you gain more practical experience with this > wait-mod/moderator feature please feel free to come back and add any useful > tidbits to the wiki. > > -MC > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/221c731a/attachment.html From saigop at gmail.com Fri May 29 02:37:21 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Fri, 29 May 2009 15:07:21 +0530 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <191c3a030905280821j1807e251ha6e5090f8fa9eeab@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> <191c3a030905270601l5d451e61y1c09afd910cb86e5@mail.gmail.com> <2ea4d47e0905280003u5ac692f3h4956db2a1ead20af@mail.gmail.com> <191c3a030905280821j1807e251ha6e5090f8fa9eeab@mail.gmail.com> Message-ID: <2ea4d47e0905290237q43ac57f0sc3648c57b757b18e@mail.gmail.com> Sorry for my presentation. Call - is nothing but the outbound number in the far end. (this is the leg I am trying to transfer) Extension - is nothing but the internal softphones My internal extension are not hanging up. On Thu, May 28, 2009 at 8:51 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That was pretty much a repeat of the same explanation. > > I am still not sure what you mean? > > What is the "call" and what is the "extension" and what is not hanging up? > > > > > On Thu, May 28, 2009 at 2:03 AM, Gopalakrishnan A.N wrote: > >> Let me explain in different way. >> As you suggested I tried the sessionX.setAutoHangup(0); and it works >> fine. I can able to transfer the call by uuid_transfer by which the call is >> originated with javascript via event socket. >> But the problem is once the call gets transferred, the extension has >> to hangup automatically rite? Thats not happening. >> >> >> >> On Wed, May 27, 2009 at 6:31 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> I am not sure what you mean at this point. >>> >>> >>> >>> On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N wrote: >>> >>>> Hi Anthony, >>>> thanks, it seems to be working, but the extension is not hanging up >>>> once I transfered the call to another mobile or to a conference. Some where >>>> I am wrong? >>>> >>>> On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> This one happens to every new guy trying to make FS into a dialer app >>>>> using JS. >>>>> >>>>> for every sessionX you create in js with the new Session constructor >>>>> >>>>> sessionX.setAutoHangup(0); >>>>> >>>>> This allows the channels to remain alive outside the script once they >>>>> are hungup/transferred etc. >>>>> >>>>> >>>>> On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N wrote: >>>>> >>>>>> Hi, >>>>>> I had some discussion with the IRC regarding the uuid_transfer gets >>>>>> hang-up where the call is originated via javascript thru event socket. I was >>>>>> suggested to install latest SVN trunk. I did that again the same issue, the >>>>>> log is attached with here http://pastebin.freeswitch.org/9103 >>>>>> >>>>>> My call flow like this, >>>>>> >>>>>> 1. api jsrun fils.js >>>>>> 2. capture the uuid >>>>>> 3. api uuid_transfer -both >>>>>> >>>>>> Both the leg gets hangedup. >>>>>> >>>>>> Someone can assist me where I am wrong. >>>>>> >>>>>> -- >>>>>> Thank you with regards, >>>>>> Gopal, >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> iax:guest at conference.freeswitch.org/888 >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Thank you with regards, >>>> Gopal, >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Thank you with regards, >> Gopal, >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/67d8205f/attachment-0001.html From saigop at gmail.com Fri May 29 02:42:53 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Fri, 29 May 2009 15:12:53 +0530 Subject: [Freeswitch-users] Freeswitch with APR In-Reply-To: References: <2ea4d47e0905280714t4b539e69w70feb7d8642d2972@mail.gmail.com> Message-ID: <2ea4d47e0905290242h7a6992f6o7f7232b522e53df8@mail.gmail.com> Ok Thanks Mike. I hope in asterisk it is not there. I was trying some couple of things, 1. when I use Java servlet to dial a call thru asterisk using manager interface it slows down. 2. when I use the same Java servlet with Freeswitch I feel the speed and usage in the tomcat. Its bit faster for simultaneous call. whereas in Asterisk I am not able to see the speed. On Thu, May 28, 2009 at 9:55 PM, Michael Jerris wrote: > apr is used primarily for portability and utility, not performance. > It is used heavily throughout the entire codebase in the core and all > modules (via our abstraction layer). > > On May 28, 2009, at 10:14 AM, Gopalakrishnan A.N wrote: > > > Hi, > > > > I saw the apache portable runtime is included in freeswitch. So > > far I understand that using APR will give good performance. Am I > > correct? or it has been used for some other scenarios like even > > socket or dialplan? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/bb214a9a/attachment.html From jason at jasonjgw.net Fri May 29 03:02:30 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 29 May 2009 20:02:30 +1000 Subject: [Freeswitch-users] ZRTP errors in logs - are they significant? Message-ID: <20090529100230.GA32337@jdc.jasonjgw.net> After ZRTP negotiation is complete (the ZRTP state machine has entered the "secure" state), I get a number of lines in the log as follows (FreeSWITCH rev. 13501): 2009-05-29 16:43:19 [DEBUG] switch_rtp.c:538 zrtp_logger() [zrtp protoco]: ERROR! Decrypt failed. ID=14:DH s=SRTP authentication failure (RTP size=172 ssrc=2760088424 seq=53043/53043 pt=9) They all have a time stamp of shortly after the end of the protocol negotiation. Is this normal, or a sign of trouble? The call itself is fine, and ZRTP does not leave the secure state until the end of the call: 2009-05-29 16:43:30 [DEBUG] switch_rtp.c:538 zrtp_logger() [ zrtp engine]: STOP STREAM ID=14 mode=DH state=SECURE. Are others seeing similar behaviour? From peter.olsson at visionutveckling.se Fri May 29 03:31:15 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 12:31:15 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn't upgraded until today, so I don't know exactly when this happened. I've noticed quite a few changes on the RTP stack, beacuse of the implementation om ZRTP, and I guess it's somewhere around this time when it happened. How to continue debugging on this issue? I have both a working version of FS (compiled 5 days ago), and a broken one (compiled today), so I can test this very easily, and everything is on a non live server. The conf-dir is the same between the revisions. The calls I'm trying to do is both directly to FS (voicemail or similar applications), and aslo calls to another SIP-trunk, to PSTN (media stream is sent through FS). Regards, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/0188fc26/attachment.html From peter.olsson at visionutveckling.se Fri May 29 03:40:59 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 12:40:59 +0200 Subject: [Freeswitch-users] FW: Something broken in RTP in latest trunk (r13502)? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C05@cooper> Sorry for missing this in my last post, but I'm using sofia for all calls. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Olsson Skickat: den 29 maj 2009 12:31 Till: 'freeswitch-users at lists.freeswitch.org' ?mne: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn't upgraded until today, so I don't know exactly when this happened. I've noticed quite a few changes on the RTP stack, beacuse of the implementation om ZRTP, and I guess it's somewhere around this time when it happened. How to continue debugging on this issue? I have both a working version of FS (compiled 5 days ago), and a broken one (compiled today), so I can test this very easily, and everything is on a non live server. The conf-dir is the same between the revisions. The calls I'm trying to do is both directly to FS (voicemail or similar applications), and aslo calls to another SIP-trunk, to PSTN (media stream is sent through FS). Regards, Peter !DSPAM:4a1fbb5432933803719169! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/8b887be0/attachment.html From jason at jasonjgw.net Fri May 29 03:46:39 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 29 May 2009 20:46:39 +1000 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> Message-ID: <20090529104639.GA2159@jdc.jasonjgw.net> Peter Olsson wrote: > After using the latest trunk revisions I get no audio anymore. The last > working build I have is about 5 days ago. I havn't upgraded until today, so > I don't know exactly when this happened. You could always check out some intermediate revisions, compile them, and see if they work. This will enable you to find out exactly which revision introduced the problem. The correct way to do this is as a binary search. In the git revision control system, this is partially automated as the bisect command. Even though FreeSWITCH uses Subversion rather than Git, you can still perform the bisect manually by checking out particular revisions by number. svn update -r should do it. From dftoro at yahoo.com Fri May 29 05:49:27 2009 From: dftoro at yahoo.com (Diego Toro) Date: Fri, 29 May 2009 05:49:27 -0700 (PDT) Subject: [Freeswitch-users] The calls are dropped during register Message-ID: <712244.11469.qm@web33503.mail.mud.yahoo.com> Hi, my job with FS has been on Windows. ? Diego --- On Thu, 5/28/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] The calls are dropped during register To: freeswitch-users at lists.freeswitch.org Date: Thursday, May 28, 2009, 8:22 PM Anything on linux? /b On May 28, 2009, at 8:19 PM, Diego Toro wrote: thank you Brian, ? my notes on?http://jira.freeswitch.org/browse/SFSIP-143, I have hardware avaible for testing Diego Brian West brian at freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/ae6c9e87/attachment-0001.html From Prometheus001 at gmx.net Fri May 29 05:58:24 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 29 May 2009 14:58:24 +0200 Subject: [Freeswitch-users] The calls are dropped during register In-Reply-To: <712244.11469.qm@web33503.mail.mud.yahoo.com> References: <712244.11469.qm@web33503.mail.mud.yahoo.com> Message-ID: <4A1FDBF0.8080900@gmx.net> And mine with the same behaviour on Linux. Best regards Peter Diego Toro schrieb: > Hi, my job with FS has been on Windows. > > Diego > > --- On *Thu, 5/28/09, Brian West //* wrote: > > > From: Brian West > Subject: Re: [Freeswitch-users] The calls are dropped during register > To: freeswitch-users at lists.freeswitch.org > Date: Thursday, May 28, 2009, 8:22 PM > > Anything on linux? > /b > > On May 28, 2009, at 8:19 PM, Diego Toro wrote: > >> thank you Brian, >> >> my notes on http://jira.freeswitch.org/browse/SFSIP-143, I have >> hardware avaible for testing >> Diego >> > > Brian West > brian at freeswitch.org > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri May 29 06:08:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 May 2009 08:08:56 -0500 Subject: [Freeswitch-users] Contact Header In-Reply-To: References: Message-ID: <191c3a030905290608p69634f6cjf4732d1ff8331652@mail.gmail.com> try enabling the Path header too we fully support that On Fri, May 29, 2009 at 1:37 AM, Jim Burke wrote: > Thanks Brian, will check it out. > > I am using FS as Voicemail behind Opensips. As we have 2 Opensips > servers if FS responds with a Contact header with a URI value we > cannot route the call back to the correct FS server and the call is > eventually dropped. For some reason this occurs even though we have > record_route set in Opensips. > > Regards, > > > On Fri, May 29, 2009 at 3:25 PM, Brian West wrote: > > This behavior only takes place on > value="true"/>, Which Is required in some cases for things like call > pickup > > and barge when using TLS/TCP. Can you tell me why the contact with the > > hostname is giving you a problem? > > /b > > On May 28, 2009, at 11:40 PM, Jim Burke wrote: > > > > Have looked at Jira and in old emails from the list and could not find > > anything, was there a variable that needs to added to sofia config to > > make sure the Contact header has an IP address instead of a URI? > > > > Brian West > > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/f8925592/attachment.html From anthony.minessale at gmail.com Fri May 29 06:12:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 May 2009 08:12:39 -0500 Subject: [Freeswitch-users] Freeswitch with APR In-Reply-To: <2ea4d47e0905290242h7a6992f6o7f7232b522e53df8@mail.gmail.com> References: <2ea4d47e0905280714t4b539e69w70feb7d8642d2972@mail.gmail.com> <2ea4d47e0905290242h7a6992f6o7f7232b522e53df8@mail.gmail.com> Message-ID: <191c3a030905290612l2e2cde4aq367e011e5044d864@mail.gmail.com> There is a very long explanation as to the differences behind asterisk AMI and FreeSWITCH Event Socket that I will not get into now but it's not related to APR or TCP socket performance whatsoever it's more about asynchronous versus monolithic modeling. On Fri, May 29, 2009 at 4:42 AM, Gopalakrishnan A.N wrote: > Ok Thanks Mike. > I hope in asterisk it is not there. I was trying some couple of things, > > 1. when I use Java servlet to dial a call thru asterisk using manager > interface it slows down. > > 2. when I use the same Java servlet with Freeswitch I feel the speed and > usage in the tomcat. Its bit faster for simultaneous call. whereas in > Asterisk I am not able to see the speed. > > > On Thu, May 28, 2009 at 9:55 PM, Michael Jerris wrote: > >> apr is used primarily for portability and utility, not performance. >> It is used heavily throughout the entire codebase in the core and all >> modules (via our abstraction layer). >> >> On May 28, 2009, at 10:14 AM, Gopalakrishnan A.N wrote: >> >> > Hi, >> > >> > I saw the apache portable runtime is included in freeswitch. So >> > far I understand that using APR will give good performance. Am I >> > correct? or it has been used for some other scenarios like even >> > socket or dialplan? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/715e9d6a/attachment.html From anthony.minessale at gmail.com Fri May 29 06:15:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 May 2009 08:15:11 -0500 Subject: [Freeswitch-users] uuid_transfer gets break In-Reply-To: <2ea4d47e0905290237q43ac57f0sc3648c57b757b18e@mail.gmail.com> References: <2ea4d47e0905250427i249a5c01qa7fb670f1c546b99@mail.gmail.com> <191c3a030905260558o3e05cfabr18772a5ccc083030@mail.gmail.com> <2ea4d47e0905270353x7d0070a4oefa9d07193ee81a7@mail.gmail.com> <191c3a030905270601l5d451e61y1c09afd910cb86e5@mail.gmail.com> <2ea4d47e0905280003u5ac692f3h4956db2a1ead20af@mail.gmail.com> <191c3a030905280821j1807e251ha6e5090f8fa9eeab@mail.gmail.com> <2ea4d47e0905290237q43ac57f0sc3648c57b757b18e@mail.gmail.com> Message-ID: <191c3a030905290615x16f1872crb5329841f3b92a4a@mail.gmail.com> maybe you have a nat issue sending byes to the phones enable debug log by pressing f8 or typing console loglevel debug and turn on sofia trace with "sofia profile internal siptrace on" capture the entire thing and paste it to http://pastebin.freeswitch.org On Fri, May 29, 2009 at 4:37 AM, Gopalakrishnan A.N wrote: > Sorry for my presentation. > Call - is nothing but the outbound number in the far end. (this is the leg > I am trying to transfer) > Extension - is nothing but the internal softphones > My internal extension are not hanging up. > > > On Thu, May 28, 2009 at 8:51 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> That was pretty much a repeat of the same explanation. >> >> I am still not sure what you mean? >> >> What is the "call" and what is the "extension" and what is not hanging up? >> >> >> >> >> On Thu, May 28, 2009 at 2:03 AM, Gopalakrishnan A.N wrote: >> >>> Let me explain in different way. >>> As you suggested I tried the sessionX.setAutoHangup(0); and it works >>> fine. I can able to transfer the call by uuid_transfer by which the call is >>> originated with javascript via event socket. >>> But the problem is once the call gets transferred, the extension has >>> to hangup automatically rite? Thats not happening. >>> >>> >>> >>> On Wed, May 27, 2009 at 6:31 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> I am not sure what you mean at this point. >>>> >>>> >>>> >>>> On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N wrote: >>>> >>>>> Hi Anthony, >>>>> thanks, it seems to be working, but the extension is not hanging up >>>>> once I transfered the call to another mobile or to a conference. Some where >>>>> I am wrong? >>>>> >>>>> On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> This one happens to every new guy trying to make FS into a dialer app >>>>>> using JS. >>>>>> >>>>>> for every sessionX you create in js with the new Session constructor >>>>>> >>>>>> sessionX.setAutoHangup(0); >>>>>> >>>>>> This allows the channels to remain alive outside the script once they >>>>>> are hungup/transferred etc. >>>>>> >>>>>> >>>>>> On Mon, May 25, 2009 at 6:27 AM, Gopalakrishnan A.N >>>>> > wrote: >>>>>> >>>>>>> Hi, >>>>>>> I had some discussion with the IRC regarding the uuid_transfer gets >>>>>>> hang-up where the call is originated via javascript thru event socket. I was >>>>>>> suggested to install latest SVN trunk. I did that again the same issue, the >>>>>>> log is attached with here http://pastebin.freeswitch.org/9103 >>>>>>> >>>>>>> My call flow like this, >>>>>>> >>>>>>> 1. api jsrun fils.js >>>>>>> 2. capture the uuid >>>>>>> 3. api uuid_transfer -both >>>>>>> >>>>>>> Both the leg gets hangedup. >>>>>>> >>>>>>> Someone can assist me where I am wrong. >>>>>>> >>>>>>> -- >>>>>>> Thank you with regards, >>>>>>> Gopal, >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Thank you with regards, >>>>> Gopal, >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Thank you with regards, >>> Gopal, >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/eeb32ee1/attachment-0001.html From brian at freeswitch.org Fri May 29 06:15:59 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 08:15:59 -0500 Subject: [Freeswitch-users] ZRTP errors in logs - are they significant? In-Reply-To: <20090529100230.GA32337@jdc.jasonjgw.net> References: <20090529100230.GA32337@jdc.jasonjgw.net> Message-ID: <9B2C76CC-CFC0-4F43-94B8-A4F077B187E5@freeswitch.org> This is normal because the switch from clear to secure can happen quickly on one end or the other and you'll have a few packets that get thru before one end is ready... nothing to be worried about. /b On May 29, 2009, at 5:02 AM, Jason White wrote: > After ZRTP negotiation is complete (the ZRTP state machine has > entered the > "secure" state), I get a number of lines in the log as follows > (FreeSWITCH > rev. 13501): > > 2009-05-29 16:43:19 [DEBUG] switch_rtp.c:538 zrtp_logger() [zrtp > protoco]: > ERROR! Decrypt failed. ID=14:DH s=SRTP authentication failure (RTP > size=172 > ssrc=2760088424 seq=53043/53043 pt=9) > > They all have a time stamp of shortly after the end of the protocol > negotiation. > > Is this normal, or a sign of trouble? > > The call itself is fine, and ZRTP does not leave the secure state > until the > end of the call: > 2009-05-29 16:43:30 [DEBUG] switch_rtp.c:538 zrtp_logger() [ zrtp > engine]: > STOP STREAM ID=14 mode=DH state=SECURE. > > Are others seeing similar behaviour? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/f1195c9f/attachment.html From brian at freeswitch.org Fri May 29 06:17:03 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 08:17:03 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <20090529104639.GA2159@jdc.jasonjgw.net> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> Message-ID: <813CCE54-A898-4694-B777-F8DC4AD99999@freeswitch.org> Its called build skew... we added an extra_data element to the frame struct. Please do a fresh checkout and build. /b On May 29, 2009, at 5:46 AM, Jason White wrote: > Peter Olsson wrote: >> After using the latest trunk revisions I get no audio anymore. The >> last >> working build I have is about 5 days ago. I havn't upgraded until >> today, so >> I don't know exactly when this happened. > > You could always check out some intermediate revisions, compile > them, and see > if they work. This will enable you to find out exactly which revision > introduced the problem. > > The correct way to do this is as a binary search. In the git > revision control > system, this is partially automated as the bisect command. Even though > FreeSWITCH uses Subversion rather than Git, you can still perform the > bisect manually by checking out particular revisions by number. > svn update -r > should do it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/d45c4e48/attachment.html From brian at freeswitch.org Fri May 29 06:18:03 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 08:18:03 -0500 Subject: [Freeswitch-users] The calls are dropped during register In-Reply-To: <4A1FDBF0.8080900@gmx.net> References: <712244.11469.qm@web33503.mail.mud.yahoo.com> <4A1FDBF0.8080900@gmx.net> Message-ID: <6F186E57-C9FA-4B6B-AC15-68C7ADD5EB2F@freeswitch.org> Peter find me on IRC and let me into your machine so I can trouble shoot this. Thanks, Brian On May 29, 2009, at 7:58 AM, Peter P GMX wrote: > And mine with the same behaviour on Linux. > Best regards > Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/90ff3728/attachment.html From anthony.minessale at gmail.com Fri May 29 06:21:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 May 2009 08:21:40 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <20090529104639.GA2159@jdc.jasonjgw.net> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> Message-ID: <191c3a030905290621q7e3d0220l31b9e535d80abca9@mail.gmail.com> the upgrade changed the switch_frame structure so most likely you did not do "make current" as we always recommend. Try that...... On Fri, May 29, 2009 at 5:46 AM, Jason White wrote: > Peter Olsson wrote: > > After using the latest trunk revisions I get no audio anymore. The last > > working build I have is about 5 days ago. I havn't upgraded until today, > so > > I don't know exactly when this happened. > > You could always check out some intermediate revisions, compile them, and > see > if they work. This will enable you to find out exactly which revision > introduced the problem. > > The correct way to do this is as a binary search. In the git revision > control > system, this is partially automated as the bisect command. Even though > FreeSWITCH uses Subversion rather than Git, you can still perform the > bisect manually by checking out particular revisions by number. > svn update -r > should do it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/77923303/attachment.html From peter.olsson at visionutveckling.se Fri May 29 06:24:17 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 15:24:17 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <20090529104639.GA2159@jdc.jasonjgw.net> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jason White Skickat: den 29 maj 2009 12:47 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Peter Olsson wrote: > After using the latest trunk revisions I get no audio anymore. The last > working build I have is about 5 days ago. I havn't upgraded until today, so > I don't know exactly when this happened. You could always check out some intermediate revisions, compile them, and see if they work. This will enable you to find out exactly which revision introduced the problem. The correct way to do this is as a binary search. In the git revision control system, this is partially automated as the bisect command. Even though FreeSWITCH uses Subversion rather than Git, you can still perform the bisect manually by checking out particular revisions by number. svn update -r should do it. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a1fbe7332937861021362! From brian at freeswitch.org Fri May 29 06:25:50 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 08:25:50 -0500 Subject: [Freeswitch-users] Contact Header In-Reply-To: References: Message-ID: <1EE33563-FD58-46FB-A9F8-1573D91A4B21@freeswitch.org> I would recommend not turning on the SLA option then... I had to add that in because when using TLS the phone would try to call the IP which would fail because the SSL cert wouldn't match and the poor phone would kill over, lock up and reboot sometimes :P GO POLYCOM! With that option not set on the profile the default/old behavior should return. /b On May 29, 2009, at 1:37 AM, Jim Burke wrote: > Thanks Brian, will check it out. > > I am using FS as Voicemail behind Opensips. As we have 2 Opensips > servers if FS responds with a Contact header with a URI value we > cannot route the call back to the correct FS server and the call is > eventually dropped. For some reason this occurs even though we have > record_route set in Opensips. > > Regards, Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/73e58c31/attachment.html From brian at freeswitch.org Fri May 29 06:26:17 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 08:26:17 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> Message-ID: <2B8344EE-50CB-4453-9A05-6A920501DC52@freeswitch.org> Nope its not a sofia issue... its build skew ;) /b On May 29, 2009, at 8:24 AM, Peter Olsson wrote: > I've looked into this a bit more now, and I think it is a sofia > issue, I will look trough the changes in sofia since I had the last > working configuration, and see if I find anything. > > /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/f86a373e/attachment.html From peter.olsson at visionutveckling.se Fri May 29 06:28:08 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 15:28:08 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <813CCE54-A898-4694-B777-F8DC4AD99999@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <813CCE54-A898-4694-B777-F8DC4AD99999@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3C@cooper> I did that, and it compiles fine. It's just not working :) But as I said in my last post, I think it could also be related to sofia, when using h323 it works... However - maybe I'm using opal's RTP stream by then..? I'll get some logs for the scenario, and if I don't find a solution I'll start a new issue on jira. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:17 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Its called build skew... we added an extra_data element to the frame struct. Please do a fresh checkout and build. /b On May 29, 2009, at 5:46 AM, Jason White wrote: Peter Olsson > wrote: After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn't upgraded until today, so I don't know exactly when this happened. You could always check out some intermediate revisions, compile them, and see if they work. This will enable you to find out exactly which revision introduced the problem. The correct way to do this is as a binary search. In the git revision control system, this is partially automated as the bisect command. Even though FreeSWITCH uses Subversion rather than Git, you can still perform the bisect manually by checking out particular revisions by number. svn update -r should do it. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a1fe21a32931540574738! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/7ef3a520/attachment.html From peter.olsson at visionutveckling.se Fri May 29 06:33:58 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 15:33:58 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <2B8344EE-50CB-4453-9A05-6A920501DC52@freeswitch.org> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> <2B8344EE-50CB-4453-9A05-6A920501DC52@freeswitch.org> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3E@cooper> Nope - it's not :) Just to make sure I even deleted the source completely, and checked everything out again. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) /b On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a1fe3bf32935202978184! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/81febe90/attachment-0001.html From anthony.minessale at gmail.com Fri May 29 06:41:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 May 2009 08:41:42 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <813CCE54-A898-4694-B777-F8DC4AD99999@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3C@cooper> Message-ID: <191c3a030905290641i2ddf289amc9144dafd152f23c@mail.gmail.com> when you say "i did that" you typed "make current" to rebuild? or you are assuming your successful compile is the same effect as cleaning the 100 object files that have the wrong data structure in them so the audio data they really seek is 8 bytes offset from where they think they are until they are deleted and recompiled by the make current command? The coincidental side-effect of this is no audio in any rtp streams...... *shrug* or you can continue to update more and more revisions on top of each other and end up with even worse build skew. We tried 3 times to tell you that's plenty...... On Fri, May 29, 2009 at 8:28 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > I did that, and it compiles fine. It?s just not working :) But as I said > in my last post, I think it could also be related to sofia, when using h323 > it works... However ? maybe I?m using opal?s RTP stream by then..? > > > > I?ll get some logs for the scenario, and if I don?t find a solution I?ll > start a new issue on jira. > > > > /Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Brian West > *Skickat:* den 29 maj 2009 15:17 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Something broken in RTP in latest trunk > (r13502)? > > > > Its called build skew... we added an extra_data element to the frame > struct. Please do a fresh checkout and build. > > > > /b > > > > On May 29, 2009, at 5:46 AM, Jason White wrote: > > > > Peter Olsson wrote: > > After using the latest trunk revisions I get no audio anymore. The last > > working build I have is about 5 days ago. I havn't upgraded until today, so > > I don't know exactly when this happened. > > > You could always check out some intermediate revisions, compile them, and > see > if they work. This will enable you to find out exactly which revision > introduced the problem. > > The correct way to do this is as a binary search. In the git revision > control > system, this is partially automated as the bisect command. Even though > FreeSWITCH uses Subversion rather than Git, you can still perform the > bisect manually by checking out particular revisions by number. > svn update -r > should do it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > !DSPAM:4a1fe21a32931540574738! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/c4540396/attachment.html From anthony.minessale at gmail.com Fri May 29 06:46:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 29 May 2009 08:46:17 -0500 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3E@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> <2B8344EE-50CB-4453-9A05-6A920501DC52@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3E@cooper> Message-ID: <191c3a030905290646g34b156aeyb8fc31f596204e6d@mail.gmail.com> did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Nope ? it?s not :) > > > > Just to make sure I even deleted the source completely, and checked > everything out again. > > > > /Peter > > > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Brian West > *Skickat:* den 29 maj 2009 15:26 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Something broken in RTP in latest trunk > (r13502)? > > > > Nope its not a sofia issue... its build skew ;) > > > > /b > > > > On May 29, 2009, at 8:24 AM, Peter Olsson wrote: > > > > I've looked into this a bit more now, and I think it is a sofia issue, I > will look trough the changes in sofia since I had the last working > configuration, and see if I find anything. > > /Peter > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > > !DSPAM:4a1fe3bf32935202978184! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/b5dd4011/attachment.html From peter.olsson at visionutveckling.se Fri May 29 06:54:09 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 15:54:09 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <191c3a030905290641i2ddf289amc9144dafd152f23c@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <813CCE54-A898-4694-B777-F8DC4AD99999@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3C@cooper> <191c3a030905290641i2ddf289amc9144dafd152f23c@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C47@cooper> Actually I deleted everything from disk and downloaded a fresh clean copy from SVN and rebuilt it from scratch. I should mention that I'm on windows, so I never do "make current". I just do a full clean, get latest from SVN and rebuild, that's what I do every time. But for this time I even deleted everything from disk and did a fresh rebuild - but nothing helped.. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:42 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? when you say "i did that" you typed "make current" to rebuild? or you are assuming your successful compile is the same effect as cleaning the 100 object files that have the wrong data structure in them so the audio data they really seek is 8 bytes offset from where they think they are until they are deleted and recompiled by the make current command? The coincidental side-effect of this is no audio in any rtp streams...... *shrug* or you can continue to update more and more revisions on top of each other and end up with even worse build skew. We tried 3 times to tell you that's plenty...... On Fri, May 29, 2009 at 8:28 AM, Peter Olsson > wrote: I did that, and it compiles fine. It's just not working :) But as I said in my last post, I think it could also be related to sofia, when using h323 it works... However - maybe I'm using opal's RTP stream by then..? I'll get some logs for the scenario, and if I don't find a solution I'll start a new issue on jira. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:17 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Its called build skew... we added an extra_data element to the frame struct. Please do a fresh checkout and build. /b On May 29, 2009, at 5:46 AM, Jason White wrote: Peter Olsson > wrote: After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn't upgraded until today, so I don't know exactly when this happened. You could always check out some intermediate revisions, compile them, and see if they work. This will enable you to find out exactly which revision introduced the problem. The correct way to do this is as a binary search. In the git revision control system, this is partially automated as the bisect command. Even though FreeSWITCH uses Subversion rather than Git, you can still perform the bisect manually by checking out particular revisions by number. svn update -r should do it. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:4a1fe7c832931532616746! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/fd43f22d/attachment-0001.html From peter.olsson at visionutveckling.se Fri May 29 06:55:24 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 29 May 2009 15:55:24 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <191c3a030905290646g34b156aeyb8fc31f596204e6d@mail.gmail.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> <2B8344EE-50CB-4453-9A05-6A920501DC52@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3E@cooper> <191c3a030905290646g34b156aeyb8fc31f596204e6d@mail.gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C48@cooper> I'm on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: Nope - it's not :) Just to make sure I even deleted the source completely, and checked everything out again. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) /b On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 !DSPAM:4a1fe8ca32931197316932! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/ffe8b11f/attachment.html From jcromes at gmail.com Fri May 29 07:22:32 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Fri, 29 May 2009 09:22:32 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> Message-ID: <4A1FEFA8.3070400@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/26a00308/attachment.html From mashudiflexi at telkom.co.id Fri May 29 07:46:33 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 29 May 2009 21:46:33 +0700 Subject: [Freeswitch-users] FS and Sangoma card got error while pickup a call Message-ID: <4A1FF549.7030200@telkom.co.id> Hi Guys, I have install WANPIPE Release: 3.5.2 for sangoma A104D and FreeSwitch 1.0.4pre8 with openzap modul. and I use Lua script for playing wav file. and I got error like this below while I call the number 0312982300, if i run ./fs_cli , the FS can pickup a call for moment, after more than 1 hour , the FS cannot give respon to any incoming call and give this error result as below. please help, thank in advance for any suggestion. mashudi 2009-05-29 21:22:14 [DEBUG] mod_openzap.c:407 channel_on_execute() OpenZAP/1:2/2982300 CHANNEL EXECUTE 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() OpenZAP/1:2/2982300 Standard EXECUTE EXECUTE OpenZAP/1:2/2982300 lua(test.lua) 2009-05-29 21:22:14 [ERR] mod_lua.cpp:182 lua_parse_and_execute() cannot open /usr/local/freeswitch/scripts/test.lua: Too many open files 2009-05-29 21:22:14 [DEBUG] switch_cpp.cpp:916 destroy() destroy/unlink session from object 2009-05-29 21:22:14 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup OpenZAP/1:2/2982300 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-29 21:22:14 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal OpenZAP/1:2/2982300 [KILL] 2009-05-29 21:22:14 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/1:2/2982300 [BREAK] 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (OpenZAP/1:2/2982300) State EXECUTE going to sleep 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/1:2/2982300) Running State Change CS_HANGUP 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (OpenZAP/1:2/2982300) State HANGUP 2009-05-29 21:22:14 [DEBUG] mod_openzap.c:472 channel_on_hangup() Changing state on 1:2 from RING to HANGUP 2009-05-29 21:22:14 [DEBUG] mod_openzap.c:485 channel_on_hangup() OpenZAP/1:2/2982300 CHANNEL HANGUP 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:2/2982300 Standard HANGUP, cause: NORMAL_CLEARING 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (OpenZAP/1:2/2982300) State HANGUP going to sleep 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (OpenZAP/1:2/2982300) State Change CS_HANGUP -> CS_REPORTING 2009-05-29 21:22:14 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal OpenZAP/1:2/2982300 [BREAK] 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (OpenZAP/1:2/2982300) Running State Change CS_REPORTING 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (OpenZAP/1:2/2982300) State REPORTING 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() OpenZAP/1:2/2982300 Standard REPORTING, cause: NORMAL_CLEARING 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (OpenZAP/1:2/2982300) State REPORTING going to sleep 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/1:2/2982300) State Change CS_REPORTING -> CS_DESTROY 2009-05-29 21:22:14 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 11 (OpenZAP/1:2/2982300) Locked, Waiting on external entities 2009-05-29 21:22:14 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 11 (OpenZAP/1:2/2982300) Ended 2009-05-29 21:22:14 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel OpenZAP/1:2/2982300 [CS_DESTROY] 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (OpenZAP/1:2/2982300) State DESTROY 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() OpenZAP/1:2/2982300 Standard DESTROY 2009-05-29 21:22:14 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (OpenZAP/1:2/2982300) State DESTROY going to sleep 2009-05-29 21:22:14 [DEBUG] ozmod_isdn.c:1032 state_advance() 1:2 STATE [HANGUP] 2009-05-29 21:22:14 [DEBUG] ozmod_isdn.c:1282 state_advance() Hangup: Direction Inbound 2009-05-29 21:22:14 [DEBUG] ozmod_isdn.c:1311 state_advance() Changing state on 1:2 from HANGUP to DOWN 2009-05-29 21:22:14 [DEBUG] ozmod_isdn.c:1032 state_advance() 1:2 STATE [DOWN] 2009-05-29 21:22:14 [DEBUG] zap_io.c:1179 zap_channel_done() channel done 1:2 2009-05-29 21:22:18 [DEBUG] ozmod_isdn.c:995 zap_isdn_921_23() READ 9 -------------------------------------------------------------------------------- [08 02 0d 47 45 08 02 82 e6] 2009-05-29 21:22:18 [DEBUG] ozmod_isdn.c:1000 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] ==================================== Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya: - hubungi 147 - http://www.telkomflexi.com - ketik INFO, sms ke 345. From gerry at pstn2.net Fri May 29 08:33:42 2009 From: gerry at pstn2.net (Gerry Hull) Date: Fri, 29 May 2009 11:33:42 -0400 Subject: [Freeswitch-users] Missing Events in mod_event_socket In-Reply-To: <191c3a030905281633s236c2ad4if9902ea97365d484@mail.gmail.com> References: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> <191c3a030905281605x45b0dad8g2840340f3482cb0@mail.gmail.com> <191c3a030905281633s236c2ad4if9902ea97365d484@mail.gmail.com> Message-ID: <98a86adf0905290833n7573b8f9m61c07cd03295b365@mail.gmail.com> Hi Anthony, I updated to rev 13496 -- now I have a different problem... I connect to the event socket interface, ask for all events... then never receive any events! >From telnet: " Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted events plain all Content-Type: command/reply Reply-Text: +OK event listener enabled plain " After this point I receive no events even though I make FS do lots of things. Am I doing something stupid, or is something broken? Gerry On Thu, May 28, 2009 at 7:33 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I dug up patch and it' clearly not the right patch and is only a self > serving kludge for jonas. > There is nothing wrong with that except he never tested our proper patch > that only has on possible problem: the timeout being too short. > > I have updated the timeout to a much higher value > > please retest revision r13496 or greater > > > > > > On Thu, May 28, 2009 at 6:05 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> and attach the patch in question >> >> >> On Thu, May 28, 2009 at 4:36 PM, Brian West wrote: >> >>> Please report bugs to http://jira.freeswitch.org >>> /b >>> >>> On May 28, 2009, at 4:20 PM, Gerry Hull wrote: >>> >>> Hello, >>> >>> I am a Windows developer who has written an application around the >>> event_socket interface. My client piece started with the C# EventSocket >>> client Jonas Gauffin had posted on CodePlex. >>> Well, Jonas did not keep up that code on Codeplex, but after >>> communicating with him, I did get the latest client-side code from the >>> freeaswitch SVN, and it seems to work fine. >>> >>> However, their is a persistent, nasty bug I'm seeing: >>> >>> On an inbound call to FreeSwitch, I get the EventChannelAnswer event, >>> which gives me some of the info I need on the incoming call. >>> Following that event, I should get an EventChannelExecuteComplete event, >>> which gives me important information like call-direction, >>> channel-state, answer-state, caller-destination-number, >>> caller-caller-id-name, etc. >>> >>> The problem I'm seeing is that EventChannelAnswer ALWAYS fires on an >>> inbound call, but EventChannelExecuteComplete does not fire --randomly. I >>> thought this mighrt have something to do with linger, >>> but executing the linger command does not help. >>> >>> Jonas made the following comment on the issue: >>> >>> "It has been a bug in the eventsocket implementation in freeswitch. It >>> can sometimes skip packets if the socket layer in the os gives an error >>> code (internal socket buffer becomes full). >>> A simple send retry usually fixes the problem. I've created a patch for >>> it long time ago (and reported it in FS jira). Mike Jerris have made an own >>> fix for the issue. I do not know if it works, I'm still >>> running my own patch. I've attached it to this email. It's a patch for >>> freeswitch\src\mod\event_handlers\mod_event_socket\ mod_event_socket.c, >>> everything works gr8 for me with it." >>> >>> Well, I have no idea how to apply the patch. >>> >>> I've downloaded the latest code from trunk at files.freeswitch.org, and >>> built FS using Visual Studio 2008. all compiles fine. However, the bug >>> sticks it's nasty head up randomly about every other call. >>> >>> I've never done a patch... I tried downloading GNU Patch for windows, and >>> tried applying it, but it reported errors. >>> >>> Has this issue been fixed in core code? If not, can someone help me >>> patch this? I'm dead in the water on a project until I resolve this. In >>> every other aspect, I've found FS to be flawless. >>> >>> Regards, >>> >>> Gerry >>> >>> >>> Brian West >>> brian at freeswitch.org >>> >>> -- Meet us at ClueCon! http://www.cluecon.com >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/6699af03/attachment-0001.html From mike at jerris.com Fri May 29 10:35:05 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 29 May 2009 13:35:05 -0400 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C48@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C00@cooper> <20090529104639.GA2159@jdc.jasonjgw.net> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C39@cooper> <2B8344EE-50CB-4453-9A05-6A920501DC52@freeswitch.org> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C3E@cooper> <191c3a030905290646g34b156aeyb8fc31f596204e6d@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB57C48@cooper> Message-ID: <60B3BDE5-1D96-4EED-B59B-A0E6E42D745F@jerris.com> Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: > I?m on Windows, so I have everything under my fs directory, but I > deleted the complete directory and did everything from scratch... > > /Peter > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] F?r Anthony Minessale > Skickat: den 29 maj 2009 15:46 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk > (r13502)? > > did you delete the binaries from /usr/local/freeswitch/bin , lib and > mod too ? > > On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: > Nope ? it?s not :) > > Just to make sure I even deleted the source completely, and checked > everything out again. > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] F?r Brian West > Skickat: den 29 maj 2009 15:26 > > > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk > (r13502)? > Nope its not a sofia issue... its build skew ;) > > On May 29, 2009, at 8:24 AM, Peter Olsson wrote: > > I've looked into this a bit more now, and I think it is a sofia > issue, I will look trough the changes in sofia since I had the last > working configuration, and see if I find anything. > > /Peter > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/6cde1263/attachment.html From rupa at rupa.com Fri May 29 13:08:11 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 29 May 2009 15:08:11 -0500 Subject: [Freeswitch-users] Connecting as a "POTS codec" to Prima LT In-Reply-To: <940350.55045.qm@web50812.mail.re2.yahoo.com> References: <940350.55045.qm@web50812.mail.re2.yahoo.com> Message-ID: On Thu, May 28, 2009 at 10:37 PM, Marc Orenberg wrote: > Hi, I'd like FreeSWITCH to be able to communicate with a Musicam "Prima LT" > device. (http://www.musicamusa.com/products/prima/PrimaLT.htm). This is a > "POTS codec", which (I've just learned) means that the connection is made > via regular POTS phone connection, but instead of transmitting voice it > transmits data packets, in this case G.722 data. > Not quite sure it is POTS though. It seems the most "normal" looking interface is ISDN BRI (1 or 2 channel). -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/edd4b9ea/attachment.html From anton at yogik.org Fri May 29 13:31:12 2009 From: anton at yogik.org (Anton Karpov) Date: Fri, 29 May 2009 15:31:12 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A1FEFA8.3070400@gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> Message-ID: <4A204610.1060805@yogik.org> I'd be very interested to see your script and dialplan , for me it's a very important issue as my conference server facing to outside and I need to have moderators and regular users entering different pins. Anton jcromes at gmail.com wrote: > Unfortunately, the instance of FreeSwitch where I've been playing with > this is at work and I can't get to it at the moment... Eventually, I > plan to post my entire implementation (javascript and dialplan) on the > Wiki because I've added quite a few improvements to the confcall.js > example that is there now... > > Here's what I did though: > > In conference.conf.xml, put this entire line into the profile you are > using for your conferences: > ** > > In your dialplan, you need to use the two lines below. Somehow, you > need to separate how callers encounter them though.. > > This first line enters callers as a standard user. This means that > anyone entering conference 1234 using this line will WAIT until a > moderator enters. > ** > > This second line sets the moderator member flag as they enter conference > 1234. Anyone already in the conference will now come off music. You > could put an extra prompt for a pass code before this line. Or you > could assign two extensions that both go to the same conference, but one > extension is publicly known, the other is only known to the owner/moderator. > * data="1234 at your-conf-profile-name+flags{moderator}"/>* > > Hope that helps. > > > Jo?o Mesquita wrote: >> I could not get this working on current trunk. Can you post your >> configuration on conference module and the dialplan example? >> >> Thanks, >> >> jmesquita > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri May 29 13:34:34 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 15:34:34 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A204610.1060805@yogik.org> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> Message-ID: <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> Maybe someone can do a wiki page with scripts and a howto? /b On May 29, 2009, at 3:31 PM, Anton Karpov wrote: > I'd be very interested to see your script and dialplan , for me it's a > very important issue as my conference server facing to outside and I > need to have moderators and regular users entering different pins. > Anton Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/066fcc0e/attachment.html From jcromes at gmail.com Fri May 29 15:23:32 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Fri, 29 May 2009 17:23:32 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> Message-ID: <4A206064.8050000@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/358ed0ba/attachment-0001.html From luismzuccolo at yahoo.com.ar Fri May 29 18:58:24 2009 From: luismzuccolo at yahoo.com.ar (Luis M. Zuccolo) Date: Fri, 29 May 2009 22:58:24 -0300 Subject: [Freeswitch-users] Error sending mail Message-ID: <1243648704.3652.17.camel@localhost.localdomain> Hi: I get this error when voicemail try to send an email: '/bin/cat /tmp/mail.12436473394319 | sendmail -t (null)' This is the called extension: Why the vm_mailto variable isnt't passed to the script? What's wrong? Someone can assist me? Thanks in advance Luis Zuccolo __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ?gratis! ?Abr? tu cuenta ya! - http://correo.yahoo.com.ar From brian at freeswitch.org Fri May 29 19:36:23 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 21:36:23 -0500 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <1243648704.3652.17.camel@localhost.localdomain> References: <1243648704.3652.17.camel@localhost.localdomain> Message-ID: Are you really using sendmail or are you using something like exim? /b On May 29, 2009, at 8:58 PM, Luis M. Zuccolo wrote: > Why the vm_mailto variable isnt't passed to the script? > What's wrong? > Someone can assist me? > > Thanks in advance > Luis Zuccolo Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090529/3f707729/attachment.html From jason at jasonjgw.net Fri May 29 20:31:17 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 30 May 2009 13:31:17 +1000 Subject: [Freeswitch-users] ZRTP errors in logs - are they significant? In-Reply-To: <9B2C76CC-CFC0-4F43-94B8-A4F077B187E5@freeswitch.org> References: <20090529100230.GA32337@jdc.jasonjgw.net> <9B2C76CC-CFC0-4F43-94B8-A4F077B187E5@freeswitch.org> Message-ID: <20090530033117.GA11634@jdc.jasonjgw.net> Brian West wrote: > This is normal because the switch from clear to secure can happen > quickly on one end or the other and you'll have a few packets that get > thru before one end is ready... nothing to be worried about. I thought that might be the scenario. In a typical FreeSWITCH to FreeSWITCH call, the entire ZRTP key negotiation takes place during early media, hence not even the first few seconds of conversation are transmitted in the clear. From brian at freeswitch.org Fri May 29 20:34:57 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 29 May 2009 22:34:57 -0500 Subject: [Freeswitch-users] ZRTP errors in logs - are they significant? In-Reply-To: <20090530033117.GA11634@jdc.jasonjgw.net> References: <20090529100230.GA32337@jdc.jasonjgw.net> <9B2C76CC-CFC0-4F43-94B8-A4F077B187E5@freeswitch.org> <20090530033117.GA11634@jdc.jasonjgw.net> Message-ID: If you happen to have a polycom or snom and you use the new sched_api extension I added to trunk (commented out) it will sched_api and snag the zrtp sas1 and sas2 strings and 4 seconds after the call is up update the display of the polycom with those two strings... kinda handy eh? For those not wanting to go hunting for it. :P Try that out! /b On May 29, 2009, at 10:31 PM, Jason White wrote: > I thought that might be the scenario. > > In a typical FreeSWITCH to FreeSWITCH call, the entire ZRTP key > negotiation > takes place during early media, hence not even the first few seconds > of > conversation are transmitted in the clear. Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From luismzuccolo at yahoo.com.ar Fri May 29 20:58:12 2009 From: luismzuccolo at yahoo.com.ar (Luis M. Zuccolo) Date: Sat, 30 May 2009 00:58:12 -0300 Subject: [Freeswitch-users] Error sending mail In-Reply-To: References: <1243648704.3652.17.camel@localhost.localdomain> Message-ID: <1243655892.3652.21.camel@localhost.localdomain> I'm using postfix, that has a compatiblilty interface to sendmail. On Fri, 2009-05-29 at 21:36 -0500, Brian West wrote: > Are you really using sendmail or are you using something like exim? > > > /b > > On May 29, 2009, at 8:58 PM, Luis M. Zuccolo wrote: > > > Why the vm_mailto variable isnt't passed to the script? > > What's wrong? > > Someone can assist me? > > > > Thanks in advance > > Luis Zuccolo > > Brian West > brian at freeswitch.org > > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org __________________________________________________ Pregunt?. Respond?. Descubr?. Todo lo que quer?as saber, y lo que ni imaginabas, est? en Yahoo! Respuestas (Beta). ?Probalo ya! http://www.yahoo.com.ar/respuestas From jason at jasonjgw.net Fri May 29 21:13:07 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 30 May 2009 14:13:07 +1000 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <1243655892.3652.21.camel@localhost.localdomain> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> Message-ID: <20090530041307.GA15924@jdc.jasonjgw.net> Luis M. Zuccolo wrote: > I'm using postfix, that has a compatiblilty interface to sendmail. I've used this with Sendmail successfully; it should work with Postfix too. See the mailer-ap and mailer-app-args variables in autoload_configs/switch.conf.xml and be sure they are set correctly for your installation. Try running the Postfix sendmail program manually to be sure that it is working correctly. My basic point is that there's something wrong in your configuration which needs to be fixed, and the best way to do that is to check everything carefully. From jason at jasonjgw.net Fri May 29 21:22:04 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 30 May 2009 14:22:04 +1000 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <20090530041307.GA15924@jdc.jasonjgw.net> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> Message-ID: <20090530042204.GA19692@jdc.jasonjgw.net> Jason White wrote: > See the mailer-ap and mailer-app-args variables in > autoload_configs/switch.conf.xml and be sure they are set correctly for your > installation. Try running the Postfix sendmail program manually to be sure > that it is working correctly. sendmail -t is the default, thus I would suggest testing that first with your Postfix sendmail compatibility script, as in: sendmail -t From: My name To: Recipient Name Subject: Test message This is a test message for the sendmail -t option. End with ctrl-d, and of course, substitute real e-mail addresses for the examples. Your Postfix logs might also reveal where the problems are. From luismzuccolo at yahoo.com.ar Fri May 29 22:25:52 2009 From: luismzuccolo at yahoo.com.ar (Luis M. Zuccolo) Date: Sat, 30 May 2009 02:25:52 -0300 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <20090530042204.GA19692@jdc.jasonjgw.net> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> Message-ID: <1243661152.3674.11.camel@localhost.localdomain> Yes, in console works well (without null). This variables are sets in switch.conf.xml: Previously I've omitted the error: sh: -c: line 0: syntax error near unexpected token `(' sh: -c: line 0: `/bin/cat /tmp/mail.124363022502bd | sendmail -t (null)' The problem is the "null" section Thanks in advance On Sat, 2009-05-30 at 14:22 +1000, Jason White wrote: > Jason White wrote: > > > See the mailer-ap and mailer-app-args variables in > > autoload_configs/switch.conf.xml and be sure they are set correctly for your > > installation. Try running the Postfix sendmail program manually to be sure > > that it is working correctly. > > sendmail -t is the default, thus I would suggest testing that first with your > Postfix sendmail compatibility script, as in: > > sendmail -t > From: My name > To: Recipient Name > Subject: Test message > > This is a test message for the sendmail -t option. > > End with ctrl-d, and of course, substitute real e-mail addresses for the > examples. > > Your Postfix logs might also reveal where the problems are. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ?gratis! ?Abr? tu cuenta ya! - http://correo.yahoo.com.ar From jcromes at gmail.com Fri May 29 23:11:43 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Sat, 30 May 2009 01:11:43 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A206064.8050000@gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> Message-ID: <4A20CE1F.40809@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/398f8609/attachment-0001.html From jason at jasonjgw.net Fri May 29 23:44:16 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 30 May 2009 16:44:16 +1000 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <1243661152.3674.11.camel@localhost.localdomain> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> Message-ID: <20090530064416.GA5051@jdc.jasonjgw.net> Luis M. Zuccolo wrote: > The problem is the "null" section Yes, switch_simple_email is probably being called with a null first argument. This shouldn't happen. Which svn revision are you on? Does it still happen with the latest svn revision? From peter.olsson at visionutveckling.se Fri May 29 23:58:25 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 30 May 2009 08:58:25 +0200 Subject: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Michael Jerris [mike at jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I?m on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: Nope ? it?s not :) Just to make sure I even deleted the source completely, and checked everything out again. Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682! From peter.olsson at visionutveckling.se Sat May 30 00:00:55 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 30 May 2009 09:00:55 +0200 Subject: [Freeswitch-users] FW: Something broken in RTP in latest trunk (r13502)? In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15A@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15C@cooper> And just to be clear, even though media flows in one direction (from FS to phone), I get no audio at all. And by the way, I mean SVN, not SNV :) Sorry for double posting... /Peter ________________________________________ Fr?n: Peter Olsson Skickat: den 30 maj 2009 08:58 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Michael Jerris [mike at jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I?m on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson > wrote: Nope ? it?s not :) Just to make sure I even deleted the source completely, and checked everything out again. Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682! From ssa1357 at yahoo.com Sat May 30 00:32:02 2009 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Sat, 30 May 2009 00:32:02 -0700 (PDT) Subject: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal In-Reply-To: References: Message-ID: <278503.58863.qm@web53403.mail.re2.yahoo.com> Dear Brian, I had compile it with SVN version and got same error. Sincerely, Sadajd ********************************************************************************************************** Message: 3 Date: Wed, 27 May 2009 08:37:23 -0500 From: Brian West Subject: Re: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset="us-ascii" You have to use the SVN version of both ptlib and OPAL and it will compile. /b On May 27, 2009, at 3:26 AM, Sadjad Seyed-Ahmadian wrote: > I faced a problem when I want to compile freeswitch with mod_opal. > It gives me a compilation error like bellow > > I used ptlib-2.6.2 and opal-3.6.2. > > Would someone please help me? > > Sincerely, > Sadjad Brian West brian at freeswitch.org *********************************************************************************************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/ebab4979/attachment.html From luismzuccolo at yahoo.com.ar Sat May 30 00:34:02 2009 From: luismzuccolo at yahoo.com.ar (Luis M. Zuccolo) Date: Sat, 30 May 2009 04:34:02 -0300 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <20090530064416.GA5051@jdc.jasonjgw.net> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> Message-ID: <1243668842.3674.13.camel@localhost.localdomain> 1.0.4pre8 On Sat, 2009-05-30 at 16:44 +1000, Jason White wrote: > Luis M. Zuccolo wrote: > > > The problem is the "null" section > > Yes, switch_simple_email is probably being called with a null first argument. > This shouldn't happen. > > Which svn revision are you on? Does it still happen with the latest svn > revision? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ?gratis! ?Abr? tu cuenta ya! - http://correo.yahoo.com.ar From peter.olsson at visionutveckling.se Sat May 30 00:43:17 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 30 May 2009 09:43:17 +0200 Subject: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal In-Reply-To: <278503.58863.qm@web53403.mail.re2.yahoo.com> References: , <278503.58863.qm@web53403.mail.re2.yahoo.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C4BBEB7E15D@cooper> There are some problems with latest trunk of opal, it's not compatible with FS anymore. Read more info on jira case MODOPAL-10. I recommend using an older revision of opal/ptlib (I'm testing r22623 right now). It also works better then using the stable versions of opal/ptlib. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Sadjad Seyed-Ahmadian [ssa1357 at yahoo.com] Skickat: den 30 maj 2009 09:32 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal Dear Brian, I had compile it with SVN version and got same error. Sincerely, Sadajd ********************************************************************************************************** Message: 3 Date: Wed, 27 May 2009 08:37:23 -0500 From: Brian West > Subject: Re: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal To: freeswitch-users at lists.freeswitch.org Message-ID: > Content-Type: text/plain; charset="us-ascii" You have to use the SVN version of both ptlib and OPAL and it will compile. /b On May 27, 2009, at 3:26 AM, Sadjad Seyed-Ahmadian wrote: > I faced a problem when I want to compile freeswitch with mod_opal. > It gives me a compilation error like bellow > > I used ptlib-2.6.2 and opal-3.6.2. > > Would someone please help me? > > Sincerely, > Sadjad Brian West brian at freeswitch.org *********************************************************************************************************************** !DSPAM:4a20e32a32935462813196! From jason at jasonjgw.net Sat May 30 00:49:31 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 30 May 2009 17:49:31 +1000 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <1243668842.3674.13.camel@localhost.localdomain> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> <1243668842.3674.13.camel@localhost.localdomain> Message-ID: <20090530074931.GA7688@jdc.jasonjgw.net> Luis M. Zuccolo wrote: > 1.0.4pre8 It works for me with revision 13501. Mine is later than yours. Try upgrading. From Prometheus001 at gmx.net Sat May 30 04:03:22 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 30 May 2009 13:03:22 +0200 Subject: [Freeswitch-users] xmpp Message-ID: <4A21127A.2000006@gmx.net> I saw that xmpp is supported in Fresswitch. See wiki: http://wiki.freeswitch.org/wiki/Mod_xmpp_event Has anybody already set this up? I have found no mod_xmpp neither in my mod directory nor in the source? There was also a question: " Q: Is it possible to send commands to fs via xmpp? Answer: Yes. " Anybody knows what can be done here and how to do this? Best regards Peter From Prometheus001 at gmx.net Sat May 30 04:27:50 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 30 May 2009 13:27:50 +0200 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <20090530074931.GA7688@jdc.jasonjgw.net> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> <1243668842.3674.13.camel@localhost.localdomain> <20090530074931.GA7688@jdc.jasonjgw.net> Message-ID: <4A211836.9050901@gmx.net> I have a problem where FS gives a core file when an voicemail email shall be sent via exim. I am on 13438. No entry in debug log in FS. No entry in exim log. Best regards Peter Jason White schrieb: > Luis M. Zuccolo wrote: > >> 1.0.4pre8 >> > > It works for me with revision 13501. Mine is later than yours. Try upgrading. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From asannucci at gmail.com Sat May 30 05:15:13 2009 From: asannucci at gmail.com (bakko) Date: Sat, 30 May 2009 14:15:13 +0200 Subject: [Freeswitch-users] xmpp In-Reply-To: <4A21127A.2000006@gmx.net> References: <4A21127A.2000006@gmx.net> Message-ID: <5DB766894EBF4C7A9D6F0CD26E32B35F@voztovoice> The xmpp module name is mod_dingaling You have to compile freeswitch wiht the support for this module. I use it with googletalk and Openfire (with openfire i have some problems to mantain the conection). > There was also a question: > " Q: Is it possible to send commands to fs via xmpp? > Answer: Yes. " I don't know. Regards From asannucci at gmail.com Sat May 30 06:15:01 2009 From: asannucci at gmail.com (bakko) Date: Sat, 30 May 2009 15:15:01 +0200 Subject: [Freeswitch-users] Openfire and FS xmpp Message-ID: <66EBF2D7510E4C03B4C864832C3578C4@voztovoice> I can't mantain a conection (client-server) with FS -> openfire server. In the console I receive this error: freeswitch at internal> 2009-05-30 08:06:32 [DEBUG] libdingaling.c:1548 xmpp_connect() io error 2 7 retry in 1 second(s) 2009-05-30 08:06:34 [DEBUG] libdingaling.c:1235 on_stream() XMPP server connected 2009-05-30 08:06:34 [DEBUG] libdingaling.c:1247 on_stream() XMPP authenticated This happens each 5/10 minuts Any idea? Thank you Regards From brian at freeswitch.org Sat May 30 08:21:13 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 30 May 2009 10:21:13 -0500 Subject: [Freeswitch-users] I have problem in compiling freeswitch with mod_opal In-Reply-To: <278503.58863.qm@web53403.mail.re2.yahoo.com> References: <278503.58863.qm@web53403.mail.re2.yahoo.com> Message-ID: <97A6BDDA-722E-4D7C-8B37-F147C914D1CB@freeswitch.org> Yah Robert recently broken API. Read MODOPAL-10 on JIRA. /b On May 30, 2009, at 2:32 AM, Sadjad Seyed-Ahmadian wrote: > Dear Brian, > > I had compile it with SVN version and got same error. > > Sincerely, > Sadajd Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/38b7649c/attachment.html From brian at freeswitch.org Sat May 30 08:21:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 30 May 2009 10:21:50 -0500 Subject: [Freeswitch-users] xmpp In-Reply-To: <4A21127A.2000006@gmx.net> References: <4A21127A.2000006@gmx.net> Message-ID: Not used any longer. /b On May 30, 2009, at 6:03 AM, Peter P GMX wrote: > Anybody knows what can be done here and how to do this? Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/977c534e/attachment.html From brian at freeswitch.org Sat May 30 08:34:05 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 30 May 2009 10:34:05 -0500 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <4A211836.9050901@gmx.net> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> <1243668842.3674.13.camel@localhost.localdomain> <20090530074931.GA7688@jdc.jasonjgw.net> <4A211836.9050901@gmx.net> Message-ID: <4687E605-DBC2-46C2-9487-026D4924E13A@freeswitch.org> Please Open a JIRA ASAP. We are working to get 1.0.4 out and these are the types of issues that should have been reported weeks ago if they were happening. /b On May 30, 2009, at 6:27 AM, Peter P GMX wrote: > I have a problem where FS gives a core file when an voicemail email > shall be sent via exim. > I am on 13438. > No entry in debug log in FS. > No entry in exim log. > > Best regards > Peter > > Jason White schrieb: >> Luis M. Zuccolo wrote: >> >>> 1.0.4pre8 >>> > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/19e860a0/attachment.html From brian at freeswitch.org Sat May 30 08:35:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 30 May 2009 10:35:30 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A20CE1F.40809@gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> Message-ID: <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> I would set the endconf flag in addition to the moderator flag so when you leave it kicks everyone. :) /b On May 30, 2009, at 1:11 AM, jcromes at gmail.com wrote: > Regarding #3: > Here is what I'm used to... When the (last) moderator drops, the > users hear something like "the moderator has left the conference" > and then get kicked back to music. If a moderator dials back in, > music goes away and the users are re-joined. It's intended to be a > secure conference, and the moderator is the security, right? Other > conferencing systems I've worked with behave like this as well (not > just Asterisk). I realize we're getting into the realm of fancy > features rather than a clean conference application, so I'm already > prepared take a hike. =) Could I emulate that behavior another way? > > Regarding #4: > I don't have an opinion either way on the ding sounds really - it's > a cosmetic thing and I just thought I'd mention it. > > Thanks! Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/9f820e58/attachment-0001.html From asannucci at gmail.com Sat May 30 08:54:46 2009 From: asannucci at gmail.com (bakko) Date: Sat, 30 May 2009 17:54:46 +0200 Subject: [Freeswitch-users] Syntax error on mysql-5.0.sql Message-ID: <086083CE47CD412C9B3077B7A44E46CC@voztovoice> I'm tryng to use mod_lcr In the mysql-5.0.sql (directory ../scripts/contrib/intralanman/C/lcr/sql) there is a sintax error in this line: `cid` varchar(32), NOT NULL DEFAULT '', Right: `cid` varchar(32) NOT NULL DEFAULT '', Lcr table Regards From jcromes at gmail.com Sat May 30 09:48:56 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Sat, 30 May 2009 11:48:56 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> Message-ID: <4A216378.9020808@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/aecb36b6/attachment.html From brian at freeswitch.org Sat May 30 10:06:45 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 30 May 2009 12:06:45 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A216378.9020808@gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> Message-ID: +flags{endconf|moderator} Its there... /b On May 30, 2009, at 11:48 AM, jcromes at gmail.com wrote: > Did someone add that feature? Are you messing with my head? =) > > That would almost work perfectly... One scenario to consider - > sometimes, in a large group of folks that meet regularly, there are > multiple callers that come in as a moderator because they're not > sure who might not be able to attend that particular conference. > So, we would only want to kick the users/members out if the > moderator "count" goes to 0. > > I'm such a pain. From jcromes at gmail.com Sat May 30 11:40:34 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Sat, 30 May 2009 13:40:34 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> Message-ID: <4A217DA2.7030402@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090530/fbf41a47/attachment.html From luismzuccolo at yahoo.com.ar Sat May 30 11:44:16 2009 From: luismzuccolo at yahoo.com.ar (Luis M. Zuccolo) Date: Sat, 30 May 2009 15:44:16 -0300 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <20090530074931.GA7688@jdc.jasonjgw.net> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> <1243668842.3674.13.camel@localhost.localdomain> <20090530074931.GA7688@jdc.jasonjgw.net> Message-ID: <1243709056.3625.2.camel@localhost.localdomain> I've upgrade to 13523 and I get the same result. sh: -c: line 0: syntax error near unexpected token `(' sh: -c: line 0: `/bin/cat /tmp/mail.1243708954adb2 | sendmail -t (null)' Thanks in advance. Luis Zuccolo On Sat, 2009-05-30 at 17:49 +1000, Jason White wrote: > Luis M. Zuccolo wrote: > > 1.0.4pre8 > > It works for me with revision 13501. Mine is later than yours. Try upgrading. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ?gratis! ?Abr? tu cuenta ya! - http://correo.yahoo.com.ar From Prometheus001 at gmx.net Sat May 30 14:08:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 30 May 2009 23:08:00 +0200 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <4687E605-DBC2-46C2-9487-026D4924E13A@freeswitch.org> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> <1243668842.3674.13.camel@localhost.localdomain> <20090530074931.GA7688@jdc.jasonjgw.net> <4A211836.9050901@gmx.net> <4687E605-DBC2-46C2-9487-026D4924E13A@freeswitch.org> Message-ID: <4A21A030.5050802@gmx.net> JIRA opened: *FSCORE-375 * Brian West schrieb: > Please Open a JIRA ASAP. We are working to get 1.0.4 out and these > are the types of issues that should have been reported weeks ago if > they were happening. > > /b > > On May 30, 2009, at 6:27 AM, Peter P GMX wrote: > >> I have a problem where FS gives a core file when an voicemail email >> shall be sent via exim. >> I am on 13438. >> No entry in debug log in FS. >> No entry in exim log. >> >> Best regards >> Peter >> >> Jason White schrieb: >>> Luis M. Zuccolo >> > wrote: >>> >>>> 1.0.4pre8 >>>> >> > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Sat May 30 17:40:34 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 31 May 2009 10:40:34 +1000 Subject: [Freeswitch-users] Error sending mail In-Reply-To: <1243709056.3625.2.camel@localhost.localdomain> References: <1243648704.3652.17.camel@localhost.localdomain> <1243655892.3652.21.camel@localhost.localdomain> <20090530041307.GA15924@jdc.jasonjgw.net> <20090530042204.GA19692@jdc.jasonjgw.net> <1243661152.3674.11.camel@localhost.localdomain> <20090530064416.GA5051@jdc.jasonjgw.net> <1243668842.3674.13.camel@localhost.localdomain> <20090530074931.GA7688@jdc.jasonjgw.net> <1243709056.3625.2.camel@localhost.localdomain> Message-ID: <20090531004034.GA13348@jdc.jasonjgw.net> Luis M. Zuccolo wrote: > I've upgrade to 13523 and I get the same result. My only suggestion at this point is to debug it with gdb to find out why a null argument is being passed to the function. There must be something in your configuration or environment that differs from mine, and which is responsible for the problem - it works fine here. From jcromes at gmail.com Sat May 30 19:16:30 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Sat, 30 May 2009 21:16:30 -0500 Subject: [Freeswitch-users] Conference users hear MOH until leader enters? In-Reply-To: <4A217DA2.7030402@gmail.com> References: <4A1BFECE.7070603@gmail.com> <2DF46C98-C6EC-4595-AD66-DC14B9FBC8D7@freeswitch.org> <4A1C0ECD.5090900@gmail.com> <191c3a030905261656q667178o726a509f13c6bf3@mail.gmail.com> <87f2f3b90905261755q2b98de83sd9683bb3465649b9@mail.gmail.com> <23754379.post@talk.nabble.com> <87f2f3b90905280856j4e6be8bcu3dddca32bfc2a60f@mail.gmail.com> <5a8712120905290018h308c70f3g6869a604a139aa6e@mail.gmail.com> <4A1FEFA8.3070400@gmail.com> <4A204610.1060805@yogik.org> <209F9859-9B7F-421E-A70A-C9A823C71C11@freeswitch.org> <4A206064.8050000@gmail.com> <4A20CE1F.40809@gmail.com> <36FC691D-93B3-4004-8706-D100D5BA672F@freeswitch.org> <4A216378.9020808@gmail.com> <4A217DA2.7030402@gmail.com> Message-ID: <4A21E87E.90709@gmail.com> I think I can answer my own question after looking at the code... It seems that when THAT ONE user leaves, a flag is set that notifies the conference thread to teardown the conference. I guess I will have to roll my own on this one I guess, especially since I don't want to kill the conference completely, just drop the users back to music. Also, more importantly... I just discovered a number of conference profile options that are neither documented in the Wiki nor mentioned in the sample configuration file. I've added entries in the Wiki for all the ones that were missing, but I don't know what half of them do. =( Could someone in-the-know please fill those in? Also, I would suggest adding those to the sample config file. Options like "endconf" and "announce-user" are GREAT conference features, but no one knows they are there! (I had actually implemented the user count announcement within Javascript, because I didn't know it was available.) From fvillarroel at yahoo.com Sat May 30 20:23:47 2009 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Sat, 30 May 2009 20:23:47 -0700 (PDT) Subject: [Freeswitch-users] g729 support Message-ID: <839380.11137.qm@web34305.mail.mud.yahoo.com> Dear All, In this moment i am using Asterisk for my services VoIP; i am testing FS and i am very interest on change Asterisk for FS, my motivation is improve my qualify mainly ACD statics. But i have problem, G729 codec is only supported in passthrough mode. FreeSwitch does not do any transcoding with G729 yet, hence I can't fully move to FS without G729 support. I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to receive this traffic and convert to ulaw and forward to other SIP provider, how i can do or anyone help me? I hope your comments or idea how i can do, Fernando. From jcromes at gmail.com Sat May 30 22:43:18 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Sun, 31 May 2009 00:43:18 -0500 Subject: [Freeswitch-users] JavaScript apiExecute vs session.execute Message-ID: <4A2218F6.2010803@gmail.com> From JavaScript, what is the difference between apiExecute and session.execute? It seems they are mostly interchangable, in fact the Wiki gives examples of them doing identical operations. What advantages does one have over the other? Are there some things that cannot or should not be called from one or the other? Thanks! From mike at jerris.com Sat May 30 23:10:01 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 31 May 2009 02:10:01 -0400 Subject: [Freeswitch-users] JavaScript apiExecute vs session.execute In-Reply-To: <4A2218F6.2010803@gmail.com> References: <4A2218F6.2010803@gmail.com> Message-ID: <520C5CF0-8E87-4560-AE75-B2863F244920@jerris.com> apiExecute is for api commands (the ones you can run at the cli) and session.execute is for applications (the ones you run from the dialplan). Mike On May 31, 2009, at 1:43 AM, jcromes at gmail.com wrote: > From JavaScript, what is the difference between apiExecute and > session.execute? > It seems they are mostly interchangable, in fact the Wiki gives > examples > of them doing identical operations. > > What advantages does one have over the other? > Are there some things that cannot or should not be called from one or > the other? > > Thanks! > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Sat May 30 23:39:46 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 30 May 2009 23:39:46 -0700 Subject: [Freeswitch-users] g729 support In-Reply-To: <839380.11137.qm@web34305.mail.mud.yahoo.com> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> Message-ID: <283A8718-83F3-456A-A2EA-B71F36CD21C2@freeswitch.org> You have a few choices. If you can hold out until g729 licensing is available in FreeSWITCH then that's a viable option. If not then you'll need to go the hardware route. Go to wiki.freeswitch.org and search for "mid_dahdi_codec" and you can learn more about the details. Essentially you can buy a card that does the g729 transcoding - license included - and still use FS. -MC Sent from my iPhone On May 30, 2009, at 8:23 PM, FERNANDO VILLARROEL wrote: > > Dear All, > > In this moment i am using Asterisk for my services VoIP; i am > testing FS and i am very interest on change Asterisk for FS, my > motivation is improve my qualify mainly ACD statics. > > But i have problem, G729 codec is only supported in passthrough > mode. FreeSwitch does not do any transcoding with G729 yet, hence I > can't fully move to FS without G729 support. > > I am receiving traffic in g729 in Asterisk(does very bad) i need a > FS to receive this traffic and convert to ulaw and forward to other > SIP provider, how i can do or anyone help me? > > I hope your comments or idea how i can do, > > Fernando. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mkezys at gmail.com Sun May 31 00:05:53 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Sun, 31 May 2009 10:05:53 +0300 Subject: [Freeswitch-users] g729 support In-Reply-To: <839380.11137.qm@web34305.mail.mud.yahoo.com> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> Message-ID: <0bfa01c9e1be$3e244510$ba6ccf30$@com> If you live in patent-free country, you can try this: http://github.com/Deepwalker/fs_itu_g729/tree/master Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of FERNANDO VILLARROEL Sent: 2009 m. gegu??s 31 d. 06:24 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] g729 support Dear All, In this moment i am using Asterisk for my services VoIP; i am testing FS and i am very interest on change Asterisk for FS, my motivation is improve my qualify mainly ACD statics. But i have problem, G729 codec is only supported in passthrough mode. FreeSwitch does not do any transcoding with G729 yet, hence I can't fully move to FS without G729 support. I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to receive this traffic and convert to ulaw and forward to other SIP provider, how i can do or anyone help me? I hope your comments or idea how i can do, Fernando. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From raffaele.p.guidi at gmail.com Sun May 31 00:45:05 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sun, 31 May 2009 09:45:05 +0200 Subject: [Freeswitch-users] g729 support In-Reply-To: <283A8718-83F3-456A-A2EA-B71F36CD21C2@freeswitch.org> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <283A8718-83F3-456A-A2EA-B71F36CD21C2@freeswitch.org> Message-ID: I've tried that search in the wiki but with no result. Can you provide the exact url? On Sun, May 31, 2009 at 08:39, Michael S Collins wrote: > You have a few choices. If you can hold out until g729 licensing is > available in FreeSWITCH then that's a viable option. If not then > you'll need to go the hardware route. Go to wiki.freeswitch.org and > search for "mid_dahdi_codec" and you can learn more about the details. > Essentially you can buy a card that does the g729 transcoding - > license included - and still use FS. > > -MC > > Sent from my iPhone > > On May 30, 2009, at 8:23 PM, FERNANDO VILLARROEL > wrote: > > > > > Dear All, > > > > In this moment i am using Asterisk for my services VoIP; i am > > testing FS and i am very interest on change Asterisk for FS, my > > motivation is improve my qualify mainly ACD statics. > > > > But i have problem, G729 codec is only supported in passthrough > > mode. FreeSwitch does not do any transcoding with G729 yet, hence I > > can't fully move to FS without G729 support. > > > > I am receiving traffic in g729 in Asterisk(does very bad) i need a > > FS to receive this traffic and convert to ulaw and forward to other > > SIP provider, how i can do or anyone help me? > > > > I hope your comments or idea how i can do, > > > > Fernando. > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/6df4a916/attachment.html From raffaele.p.guidi at gmail.com Sun May 31 00:48:13 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Sun, 31 May 2009 09:48:13 +0200 Subject: [Freeswitch-users] g729 support In-Reply-To: <0bfa01c9e1be$3e244510$ba6ccf30$@com> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <0bfa01c9e1be$3e244510$ba6ccf30$@com> Message-ID: Do you think fs_itu_g729 would work on windows, too? Has this ever been tried? 2009/5/31 Mindaugas Kezys > If you live in patent-free country, you can try this: > http://github.com/Deepwalker/fs_itu_g729/tree/master > > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > VoIP Billing and Routing Solutions > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > FERNANDO > VILLARROEL > Sent: 2009 m. gegu??s 31 d. 06:24 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] g729 support > > > Dear All, > > In this moment i am using Asterisk for my services VoIP; i am testing FS > and > i am very interest on change Asterisk for FS, my motivation is improve my > qualify mainly ACD statics. > > But i have problem, G729 codec is only supported in passthrough mode. > FreeSwitch does not do any transcoding with G729 yet, hence I can't fully > move to FS without G729 support. > > I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to > receive this traffic and convert to ulaw and forward to other SIP provider, > how i can do or anyone help me? > > I hope your comments or idea how i can do, > > Fernando. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/1cbc42d2/attachment.html From steveu at coppice.org Sun May 31 01:08:19 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 31 May 2009 16:08:19 +0800 Subject: [Freeswitch-users] g729 support In-Reply-To: <0bfa01c9e1be$3e244510$ba6ccf30$@com> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <0bfa01c9e1be$3e244510$ba6ccf30$@com> Message-ID: <4A223AF3.8020501@coppice.org> Mindaugas Kezys wrote: > If you live in patent-free country, you can try this: > http://github.com/Deepwalker/fs_itu_g729/tree/master > You live in Antarctica? :-\ Steve From dave at 3c.co.uk Sun May 31 01:54:00 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 31 May 2009 09:54:00 +0100 Subject: [Freeswitch-users] g729 support References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <0bfa01c9e1be$3e244510$ba6ccf30$@com> Message-ID: <6B5EB23E1E0C48358593C29B6056B2AA@DELL9> > If you live in patent-free country, you can try this: > http://github.com/Deepwalker/fs_itu_g729/tree/master I've a friend who says he knows of someone who's tried it in non- patent-free countries, and it works fine there too. Alternatively, an Asterisk box makes a perfectly good G.729 to G.711 transcoder. --Dave > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > VoIP Billing and Routing Solutions > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > FERNANDO > VILLARROEL > Sent: 2009 m. gegu??s 31 d. 06:24 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] g729 support > > > Dear All, > > In this moment i am using Asterisk for my services VoIP; i am testing FS > and > i am very interest on change Asterisk for FS, my motivation is improve my > qualify mainly ACD statics. > > But i have problem, G729 codec is only supported in passthrough mode. > FreeSwitch does not do any transcoding with G729 yet, hence I can't fully > move to FS without G729 support. > > I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to > receive this traffic and convert to ulaw and forward to other SIP > provider, > how i can do or anyone help me? > > I hope your comments or idea how i can do, > > Fernando. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ssa1357 at yahoo.com Sun May 31 05:23:41 2009 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Sun, 31 May 2009 05:23:41 -0700 (PDT) Subject: [Freeswitch-users] Is still Woomera supported again? In-Reply-To: References: Message-ID: <659368.557.qm@web53412.mail.re2.yahoo.com> Hi dear All, I tried to support H323 by opal thaat failed. I want to know is still Woomera supported by FreeSwitch or not. Sincerely, Sadjad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/3dd07fff/attachment.html From dujinfang at gmail.com Sun May 31 07:12:19 2009 From: dujinfang at gmail.com (dujinfang) Date: Sun, 31 May 2009 22:12:19 +0800 Subject: [Freeswitch-users] Missing Events in mod_event_socket In-Reply-To: <98a86adf0905290833n7573b8f9m61c07cd03295b365@mail.gmail.com> References: <98a86adf0905281420q2a915dcei1c5b714d55510a51@mail.gmail.com> <191c3a030905281605x45b0dad8g2840340f3482cb0@mail.gmail.com> <191c3a030905281633s236c2ad4if9902ea97365d484@mail.gmail.com> <98a86adf0905290833n7573b8f9m61c07cd03295b365@mail.gmail.com> Message-ID: On May 29, 2009, at 11:33 PM, Gerry Hull wrote: > Hi Anthony, > > I updated to rev 13496 -- now I have a different problem... I > connect to the event socket interface, ask for all events... then > never receive any events! > > From telnet: > " > Content-Type: auth/request > auth ClueCon > > Content-Type: command/reply > Reply-Text: +OK accepted > events plain all > all or ALL ? > Content-Type: command/reply > Reply-Text: +OK event listener enabled plain > " > > After this point I receive no events even though I make FS do lots > of things. > > Am I doing something stupid, or is something broken? > > Gerry > > On Thu, May 28, 2009 at 7:33 PM, Anthony Minessale > wrote: > I dug up patch and it' clearly not the right patch and is only a > self serving kludge for jonas. > There is nothing wrong with that except he never tested our proper > patch that only has on possible problem: the timeout being too short. > > I have updated the timeout to a much higher value > > please retest revision r13496 or greater > > > > > > On Thu, May 28, 2009 at 6:05 PM, Anthony Minessale > wrote: > and attach the patch in question > > > On Thu, May 28, 2009 at 4:36 PM, Brian West > wrote: > Please report bugs to http://jira.freeswitch.org > > /b > > On May 28, 2009, at 4:20 PM, Gerry Hull wrote: > >> Hello, >> >> I am a Windows developer who has written an application around the >> event_socket interface. My client piece started with the C# >> EventSocket client Jonas Gauffin had posted on CodePlex. >> Well, Jonas did not keep up that code on Codeplex, but after >> communicating with him, I did get the latest client-side code from >> the freeaswitch SVN, and it seems to work fine. >> >> However, their is a persistent, nasty bug I'm seeing: >> >> On an inbound call to FreeSwitch, I get the EventChannelAnswer >> event, which gives me some of the info I need on the incoming call. >> Following that event, I should get an EventChannelExecuteComplete >> event, which gives me important information like call-direction, >> channel-state, answer-state, caller-destination-number, caller- >> caller-id-name, etc. >> >> The problem I'm seeing is that EventChannelAnswer ALWAYS fires on >> an inbound call, but EventChannelExecuteComplete does not fire -- >> randomly. I thought this mighrt have something to do with linger, >> but executing the linger command does not help. >> >> Jonas made the following comment on the issue: >> >> "It has been a bug in the eventsocket implementation in >> freeswitch. It can sometimes skip packets if the socket layer in >> the os gives an error code (internal socket buffer becomes full). >> A simple send retry usually fixes the problem. I've created a patch >> for it long time ago (and reported it in FS jira). Mike Jerris have >> made an own fix for the issue. I do not know if it works, I'm still >> running my own patch. I've attached it to this email. It's a patch >> for freeswitch\src\mod\event_handlers\mod_event_socket\ >> mod_event_socket.c, everything works gr8 for me with it." >> >> Well, I have no idea how to apply the patch. >> >> I've downloaded the latest code from trunk at files.freeswitch.org, >> and built FS using Visual Studio 2008. all compiles fine. >> However, the bug sticks it's nasty head up randomly about every >> other call. >> >> I've never done a patch... I tried downloading GNU Patch for >> windows, and tried applying it, but it reported errors. >> >> Has this issue been fixed in core code? If not, can someone help >> me patch this? I'm dead in the water on a project until I resolve >> this. In every other aspect, I've found FS to be flawless. >> >> Regards, >> >> Gerry > > Brian West > brian at freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/81429d45/attachment.html From edpimentl at gmail.com Sun May 31 07:47:33 2009 From: edpimentl at gmail.com (EdPimentl) Date: Sun, 31 May 2009 10:47:33 -0400 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: <820833.94846.qm@web34307.mail.mud.yahoo.com> References: <820833.94846.qm@web34307.mail.mud.yahoo.com> Message-ID: <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> Please post the solution... -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/491d1630/attachment.html From mgg at giagnocavo.net Sun May 31 08:02:19 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 31 May 2009 11:02:19 -0400 Subject: [Freeswitch-users] g729 support In-Reply-To: <0bfa01c9e1be$3e244510$ba6ccf30$@com> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <0bfa01c9e1be$3e244510$ba6ccf30$@com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C670262BF8C88@mse17be1.mse17.exchange.ms> What's the difficulty in taking Digium's codec_g729a.so and writing a wrapper around it? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mindaugas Kezys Sent: Sunday, May 31, 2009 1:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] g729 support If you live in patent-free country, you can try this: http://github.com/Deepwalker/fs_itu_g729/tree/master Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of FERNANDO VILLARROEL Sent: 2009 m. gegu??s 31 d. 06:24 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] g729 support Dear All, In this moment i am using Asterisk for my services VoIP; i am testing FS and i am very interest on change Asterisk for FS, my motivation is improve my qualify mainly ACD statics. But i have problem, G729 codec is only supported in passthrough mode. FreeSwitch does not do any transcoding with G729 yet, hence I can't fully move to FS without G729 support. I am receiving traffic in g729 in Asterisk(does very bad) i need a FS to receive this traffic and convert to ulaw and forward to other SIP provider, how i can do or anyone help me? I hope your comments or idea how i can do, Fernando. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From asannucci at gmail.com Sun May 31 08:09:51 2009 From: asannucci at gmail.com (bakko) Date: Sun, 31 May 2009 17:09:51 +0200 Subject: [Freeswitch-users] FS & Wikipbx Help SOLVED In-Reply-To: <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> References: <820833.94846.qm@web34307.mail.mud.yahoo.com> <9dc4a1670905310747w1a4f97b5gdcec1c726225e4b8@mail.gmail.com> Message-ID: <99AC4244D8EB4392AD4F0DC504AFC014@voztovoice> If you understand spanish please look at: http://www.freeswitch.es/node/55 Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/1108f597/attachment.html From brian at freeswitch.org Sun May 31 08:40:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 31 May 2009 10:40:21 -0500 Subject: [Freeswitch-users] g729 support In-Reply-To: <0bfa01c9e1be$3e244510$ba6ccf30$@com> References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <0bfa01c9e1be$3e244510$ba6ccf30$@com> Message-ID: I'm going to say this one last time. DO NOT POST links to g729 on the list. You'll be happy to know that there aren't any places that the codec isn't patented even these so called patent free zones are iffy and putting the code on the list like this puts the project in a serious position. Please do not post them again. Thanks, Brian On May 31, 2009, at 2:05 AM, Mindaugas Kezys wrote: > If you live in patent-free country, you can try this: > http://github.com/Deepwalker/fs_itu_g729/tree/master > > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > VoIP Billing and Routing Solutions Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com From brian at freeswitch.org Sun May 31 08:42:04 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 31 May 2009 10:42:04 -0500 Subject: [Freeswitch-users] Is still Woomera supported again? In-Reply-To: <659368.557.qm@web53412.mail.re2.yahoo.com> References: <659368.557.qm@web53412.mail.re2.yahoo.com> Message-ID: <12BB6418-7716-4576-A1ED-C4EA5DD37AEB@freeswitch.org> Described failed? Did yo uread MODOPAL-10 on jira? /b On May 31, 2009, at 7:23 AM, Sadjad Seyed-Ahmadian wrote: > Hi dear All, > > I tried to support H323 by opal thaat failed. I want to know is > still Woomera supported by FreeSwitch or not. > > Sincerely, > Sadjad > > Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090531/790c68a1/attachment.html From jcromes at gmail.com Sun May 31 09:04:35 2009 From: jcromes at gmail.com (jcromes at gmail.com) Date: Sun, 31 May 2009 11:04:35 -0500 Subject: [Freeswitch-users] JavaScript apiExecute vs session.execute In-Reply-To: <520C5CF0-8E87-4560-AE75-B2863F244920@jerris.com> References: <4A2218F6.2010803@gmail.com> <520C5CF0-8E87-4560-AE75-B2863F244920@jerris.com> Message-ID: <4A22AA93.9000904@gmail.com> Thanks, that clears it up nicely, although I'm not sure how I didn't figure that out before. I added some tweaks to the apiExecute and session.execute pages on the Wiki for the next guy who's confused. From moises.silva at gmail.com Sun May 31 09:44:26 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 31 May 2009 11:44:26 -0500 Subject: [Freeswitch-users] g729 support In-Reply-To: References: <839380.11137.qm@web34305.mail.mud.yahoo.com> <283A8718-83F3-456A-A2EA-B71F36CD21C2@freeswitch.org> Message-ID: On Sun, May 31, 2009 at 2:45 AM, Raffaele P. Guidi wrote: > I've tried that search in the wiki but with no result. Can you provide the > exact url? > mod_dahdi_codec is already part of Freeswitch release afair, so all you need to do is compile mod_dahdi_codec and choose g729 in your codecs. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com From lcm at marshap.com Sun May 31 09:50:07 2009 From: lcm at marshap.com (Larry Marshall) Date: Sun, 31 May 2009 09:50:07 -0700 Subject: [Freeswitch-users] Help with lua and channel variable processing Message-ID: <000701c9e20f$da8759a0$8f960ce0$@com> I would like to process the ${billsec} variable in the conf/autoload_configs/cdr_csv.conf.xml file below, returning a different value for the INSERT command. I already have written a lua function which takes an integer value from ${billsec} and returns a string. It is stored in freeswitch/scripts as sec2time.lua. Do I change the sql template below to ., ${lua(billsec)}, .?