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 <gmaruzz at celliax.org>
> 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 <anthony.minessale at gmail.com> 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, <can_man at gmx.de> 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 <MSN%3Aanthony_minessale at hotmail.com>
> >
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 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
<technical at ttnc.co.uk>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 <uuid> all
>>
> "uuid_broadcast <uuid> phrase::saynumber,1" doesn't set the
> 'current_application_response' variable in the same way as "uuid_broadcast
> <uuid> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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: <FEED9321-33F1-4BC4-95EC-5BF6F69A21FF@freeswitch.org>

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




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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,  <can_man at gmx.de> 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 <gmaruzz at celliax.org>
>> 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 <anthony.minessale at gmail.com> 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, <can_man at gmx.de> 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 <MSN%3Aanthony_minessale at hotmail.com>
>> >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 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.

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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


      
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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
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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 <gmaruzz at celliax.org> 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, ?<can_man at gmx.de> 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 <gmaruzz at celliax.org>
>>> 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 <anthony.minessale at gmail.com> 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, <can_man at gmx.de> 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 <MSN%3Aanthony_minessale at hotmail.com>
>>> >
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 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 <mike at jerris.com>
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



 

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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 <gmaruzz at celliax.org> 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 <gmaruzz at celliax.org> 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, ?<can_man at gmx.de> 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 <gmaruzz at celliax.org>
>>>> 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 <anthony.minessale at gmail.com> 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, <can_man at gmx.de> 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 <MSN%3Aanthony_minessale at hotmail.com>
>>>> >
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 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 <gmaruzz at celliax.org>
> 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 <gmaruzz at celliax.org>
> 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
> <gmaruzz at celliax.org> 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, ?<can_man at gmx.de> 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 <gmaruzz at celliax.org>
> >>>> 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 <anthony.minessale at gmail.com> 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, <can_man at gmx.de> 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
> <MSN%3Aanthony_minessale at hotmail.com>
> >>>> >
> >>>>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 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 <moizchinoy at gmail.com>
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



 

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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
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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. <q.edward at gmail.com> 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




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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 <brian at freeswitch.org>
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
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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: <A2B4F48489739E439D8845296A2FD9C408BEA7@sun.Noble.local>

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

 

 

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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: <A2B4F48489739E439D8845296A2FD9C408BEA7@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BEA7@sun.Noble.local>
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
>
>
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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: <A2B4F48489739E439D8845296A2FD9C408BEA7@sun.Noble.local>
	<4256bf830905021421l11aa1a2fhe45f3efe4ad2b533@mail.gmail.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C408BEA8@sun.Noble.local>

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
<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

 

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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: <A2B4F48489739E439D8845296A2FD9C408BEA7@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BEA7@sun.Noble.local>
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

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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 <diego.viola at gmail.com> 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: <DIIE.00000A500002D31C@192.168.49.2>

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 <diego.viola at gmail.com>
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 <diego.viola at gmail.com> 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




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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




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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: <DIIE.00000A500002D31C@192.168.49.2>
References: <DIIE.00000A500002D31C@192.168.49.2>
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 <gk at exram.de> 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 <diego.viola at gmail.com>
> 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 <diego.viola at gmail.com> 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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
Message-ID: <E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>

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




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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: <E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>
References: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>
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 <brian at freeswitch.org> 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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>
	<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




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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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>
	<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 <diego.viola at gmail.com> 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 <brian at freeswitch.org> 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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>
	<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 <brian at freeswitch.org> 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
>> <mailto:Freeswitch-users at 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 <mailto:brian at freeswitch.org>
> 
> -- Meet us at ClueCon!  http://www.cluecon.com <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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<E6F42046-1935-4721-9200-8B04855F91F4@freeswitch.org>
	<86a32abc0905031417y1f676e79qdf2fc12e2808f220@mail.gmail.com>
	<86a32abc0905031426n7e2dfbe6m7aeeaac0f7a4d770@mail.gmail.com>
Message-ID: <AB10CF8C-D0CD-4BB1-BC5E-BB66A573CF54@freeswitch.org>

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




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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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
Message-ID: <ED9D86D0-0D52-4714-81EC-DC67D0F1F76B@freeswitch.org>

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




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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: <ED9D86D0-0D52-4714-81EC-DC67D0F1F76B@freeswitch.org>
References: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<ED9D86D0-0D52-4714-81EC-DC67D0F1F76B@freeswitch.org>
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 <brian at freeswitch.org> 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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<ED9D86D0-0D52-4714-81EC-DC67D0F1F76B@freeswitch.org>
	<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 <diego.viola at gmail.com> 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 <brian at freeswitch.org> 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: <DIIE.00000A500002D31C@192.168.49.2>
	<86a32abc0905031400g53c4f4cdi1a4c09ba20a7d926@mail.gmail.com>
	<ED9D86D0-0D52-4714-81EC-DC67D0F1F76B@freeswitch.org>
	<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 <diego.viola at gmail.com> wrote:
> Will post some examples on the wiki now :)
>
> Diego
>
> On Sun, May 3, 2009 at 5:32 PM, Diego Viola <diego.viola at gmail.com> 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 <brian at freeswitch.org> 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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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?

     <fifo name="sales_fifo@$${domain}" importance="0">
       <member timeout="10" simo="1"  
lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$${domain}</ 
member>
     </fifo>

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

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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: <cbbb7d4d0904302031t3a711cccy5585f4960114c2e7@mail.gmail.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>
	<cbbb7d4d0904302031t3a711cccy5585f4960114c2e7@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>	<cbbb7d4d0904302031t3a711cccy5585f4960114c2e7@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>
	<cbbb7d4d0904302031t3a711cccy5585f4960114c2e7@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>	<cbbb7d4d0904302031t3a711cccy5585f4960114c2e7@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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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 <dujinfang at gmail.com> 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?
>
>     <fifo name="sales_fifo@$${domain}" importance="0">
>       <member timeout="10" simo="1"
> lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@
> $${domain}</member>
>     </fifo>
>


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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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 <dujinfang at gmail.com> 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? 
>         
>         
>             <fifo name="sales_fifo@$${domain}" importance="0">
>               <member timeout="10" simo="1"
>         lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@
>         $${domain}</member>
>             </fifo>
>         
> 
> 
> 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
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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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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: <A148A709-126C-4074-AF94-65BCA30B95A9@gmail.com>


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 "<member timeout=x ..." in  
fifo.conf. It seems should worked as leg_timeout, but not(on my box  
using a very recent trunk version).

Thanks again.


From wchao at yahoo.com  Mon May  4 10:02:14 2009
From: wchao at yahoo.com (wchao at yahoo.com)
Date: Mon, 4 May 2009 13:02:14 -0400 (EDT)
Subject: [Freeswitch-users] Registration problem with multiple IP phones
 behind Linux NAT firewall router
Message-ID: <alpine.LRH.2.00.0905041032240.17135@osgood.microoffice.us>

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=<ip>:<port>" 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" <sip:1001 at 2.2.2.2:5060;received=2.2.2.2:5060;fs_nat=yes>
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" <sip:1002 at 1.1.1.1>;tag=807917B4-BA73B497
             SIP Display info: "Rahim Orazkuliyev"
             SIP from address: sip:1002 at 1.1.1.1
             SIP tag: 807917B4-BA73B497
         To: <sip:1002 at 1.1.1.1>
             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: <sip:1002 at 192.168.1.3>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
             Contact Binding: <sip:1002 at 192.168.1.3>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
                 URI: <sip:1002 at 192.168.1.3>
                     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" <sip:1002 at 1.1.1.1>;tag=807917B4-BA73B497
             SIP Display info: "Rahim Orazkuliyev"
             SIP from address: sip:1002 at 1.1.1.1
             SIP tag: 807917B4-BA73B497
         To: <sip:1002 at 1.1.1.1>;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" <sip:1003 at 2.2.2.2:2058;line=5gjot59h;received=2.2.2.2:2058;fs_nat=yes>
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 <brian at freeswitch.org>
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
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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




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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 <brian at freeswitch.org>
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
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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: <FB773EC5-BE7E-4416-BFFD-706584910DE7@freeswitch.org>

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




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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: <FB773EC5-BE7E-4416-BFFD-706584910DE7@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>	<8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com>
	<FB773EC5-BE7E-4416-BFFD-706584910DE7@freeswitch.org>
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 
> <mailto: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 <mailto:brian at freeswitch.org>
>
> -- Meet us at ClueCon!  http://www.cluecon.com <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: <FB773EC5-BE7E-4416-BFFD-706584910DE7@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><8CB9AF753CA48B7-1014-CC6@WEBMAIL-DY21.sysops.aol.com>
	<FB773EC5-BE7E-4416-BFFD-706584910DE7@freeswitch.org>
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 <brian at freeswitch.org>
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



 

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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 <xml>  
tag.  Apparently, the <xml> 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 <xml>
> tag.  Apparently, the <xml> 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 <xml> 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




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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 "<xml>" isn't on the first line of the file. :)

-MC

On Mon, May 4, 2009 at 11:52 AM, Jay Austad <austad at signal15.com> 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 <xml>
> tag.  Apparently, the <xml> 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
>
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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
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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: <sip:723329 at 217.xxx.xxx.186:2651>.
Max-Forwards: 69.
From: "Extension 723321" <sip:723321 at 217.xxx.xxx.190>;tag=HB02U2mHX28yK.
To:
<sip:723329 at 217.xxx.xxx.186:2651;transport=TLS;rinstance=6c215161c08f55da>.
Call-ID: e3b50c2c-b3a0-122c-4491-001e904cc34e.
CSeq: 114620396 INVITE.
Contact: <sip:mod_sofia at 217.xxx.xxx.190:5061;transport=tls>.
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"
<sip:723321 at 217.xxx.xxx.190>;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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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 <dujinfang at gmail.com> 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

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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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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 <dujinfang at gmail.com> 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?
>>
>>
>>     <fifo name="sales_fifo@$${domain}" importance="0">
>>       <member timeout="10" simo="1"  
>> lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$$ 
>> {domain}</member>
>>     </fifo>
>>
>>
>>
>> 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

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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
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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 <stevecrozz at gmail.com> 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 <jason at jasonjgw.net> wrote:

> Stephen Crosby <stevecrozz at gmail.com> 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
>
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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 <stevecrozz at gmail.com>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 <jason at jasonjgw.net> wrote:
>
>> Stephen Crosby <stevecrozz at gmail.com> 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
>>
>
>
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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 <stevecrozz at gmail.com>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 <stevecrozz at gmail.com>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 <jason at jasonjgw.net> wrote:
>>
>>> Stephen Crosby <stevecrozz at gmail.com> 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
>>>
>>
>>
>
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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 <stevecrozz at gmail.com> 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 <stevecrozz at gmail.com> 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 <jason at jasonjgw.net> wrote:

> Stephen Crosby <stevecrozz at gmail.com> 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
>
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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 <stevecrozz at gmail.com> 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>
	<BA19DA84-74F1-4825-8040-1929E519BC14@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: <D9B44E5E-5F4F-4936-9E9F-3AC2438127D9@gmail.com>

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 <dujinfang at gmail.com> 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?
>
>     <fifo name="sales_fifo@$${domain}" importance="0">
>       <member timeout="10" simo="1"  
> lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$$ 
> {domain}</member>
>     </fifo>
>
>
> 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

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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 <Prometheus001 at gmx.net> 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




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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




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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

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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




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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 <mailto:brian at freeswitch.org>
>
> -- Meet us at ClueCon!  http://www.cluecon.com <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




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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 <brian at freeswitch.org> 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
>
>
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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 <stevecrozz at gmail.com> 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 <brian at freeswitch.org> 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
>>
>>
>
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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 <stevecrozz at gmail.com> 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 <stevecrozz at gmail.com>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 <brian at freeswitch.org> 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<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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

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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 <saeedahmad1981 at gmail.com>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
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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 <saeedahmad1981 at gmail.com>
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

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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




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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
>> <mailto:context=default%7Dsofia/default/723321 at sip2.mydomain.de>;transport=tls
>
>
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
> -- Meet us at ClueCon!  http://www.cluecon.com <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




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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 <mailto:brian at freeswitch.org>
>
> -- Meet us at ClueCon!  http://www.cluecon.com <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
blog<http://cluecon.com/blog/1>on the ClueCon
website <http://www.cluecon.com>. 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
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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  
> <saeedahmad1981 at gmail.com> 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

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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: <COL119-W65919DD0EA374BAB5558C7AB660@phx.gbl>


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
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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: <d021d3be0905052336h58316540p94c58444d4360e06@mail.gmail.com>

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
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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: <EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>

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 <saeedahmad1981 at gmail.com>
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



 

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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: <d021d3be0905052336h58316540p94c58444d4360e06@mail.gmail.com>
References: <d021d3be0905052336h58316540p94c58444d4360e06@mail.gmail.com>
Message-ID: <9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org>

add <param name="auto-restart" value="false"/> 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




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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: <EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop><87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com><77308CE88F604444863741D590835B10@saeedlaptop>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
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  
> <saeedahmad1981 at gmail.com> 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

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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: <d021d3be0905052336h58316540p94c58444d4360e06@mail.gmail.com>
	<9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org>
Message-ID: <d021d3be0905060557w1db09cb3g93b7a1ca56ffbb0@mail.gmail.com>

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 <brian at freeswitch.org> wrote:

> add <param name="auto-restart" value="false"/>
> 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
>
>
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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: <d021d3be0905060557w1db09cb3g93b7a1ca56ffbb0@mail.gmail.com>
References: <d021d3be0905052336h58316540p94c58444d4360e06@mail.gmail.com>
	<9C86182B-1237-468D-AA39-4669A6DB96CF@freeswitch.org>
	<d021d3be0905060557w1db09cb3g93b7a1ca56ffbb0@mail.gmail.com>
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




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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: <COL119-W65919DD0EA374BAB5558C7AB660@phx.gbl>
References: <COL119-W65919DD0EA374BAB5558C7AB660@phx.gbl>
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>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<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 <dujinfang at gmail.com> 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<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<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 <saeedahmad1981 at gmail.com>
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.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<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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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 <http://www.freeswitch.org/node/181>.

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
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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: <COL119-W65919DD0EA374BAB5558C7AB660@phx.gbl>
	<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: <COL119-W65919DD0EA374BAB5558C7AB660@phx.gbl>
	<4A0194FE.9030609@freeswitch.org>
	<6A31DE88-C357-4167-924C-346CBCAAEAC9@gmail.com>
Message-ID: <CBA113BE-3E4B-45E1-A852-77B34A38FF78@freeswitch.org>

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




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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><EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop><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 <dujinfang at gmail.com> 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 <saeedahmad1981 at gmail.com>
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
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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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>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<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 <dujinfang at gmail.com> 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  
>> <saeedahmad1981 at gmail.com> 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

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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




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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 <brian at freeswitch.org> 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 <http://www.freeswitch.org/node/182> for you all to
check out. Please check it out and pass it on.

-Michael
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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 <msc at freeswitch.org> 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 <gmaruzz at celliax.org>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
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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 <msc at freeswitch.org> 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.
>

<blush>I was suggesting to put mod_skypiax in the
http://www.freeswitch.org/node/182 page, for ourselves </blush>

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 <msc at freeswitch.org>
To: "freeswitch-users at lists.freeswitch.org" <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




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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: <ECDE2571-92A4-43B8-874A-A8FD3E50918D@freeswitch.org>

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




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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: <ECDE2571-92A4-43B8-874A-A8FD3E50918D@freeswitch.org>
References: <1747C3A6-465A-41A2-894B-8BE528BA9728@signal15.com>
	<ECDE2571-92A4-43B8-874A-A8FD3E50918D@freeswitch.org>
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

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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 <gmaruzz at celliax.org>wrote:

> On Wed, May 6, 2009 at 8:23 PM, Michael Collins <msc at freeswitch.org>
> 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.
> >
>
> <blush>I was suggesting to put mod_skypiax in the
> http://www.freeswitch.org/node/182 page, for ourselves </blush>
>

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
>
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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><ECDE2571-92A4-43B8-874A-A8FD3E50918D@freeswitch.org>
	<2FCAD8B3-0119-42BD-9F50-60907077D337@signal15.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C408BEC9@sun.Noble.local>

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 <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

 

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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: <COL119-W65919DD0EA374BAB5558C7AB660@phx.gbl>
	<4A0194FE.9030609@freeswitch.org>
Message-ID: <COL119-W3118CBF76D2365F7C95144AB670@phx.gbl>



 

> 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
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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><EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop><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 <dujinfang at gmail.com> 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  
> <saeedahmad1981 at gmail.com> 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
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>
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>
>
>
> -- 
> 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
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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)

 

  <X-PRE-PROCESS cmd="include" data="vars.xml"/>

  #set "external_rtp_ip=174.129.201.96"  

  <section name="configuration" description="Various Configuration">

    <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>

  </section>

 

 

In conf/autoload_configs/sofia.conf.xml

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

<param name="ext-sip-ip" value="$${external_rtp_ip}"/>

 

 

Internal network IP assignment:

Name: