February 2008 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

Re: [Freeswitch-dev] VAD, TALK and NOTALK events
by Anthony Minessale 05 Mar '08

05 Mar '08

The algorithm used is not so great and could stand to be replaced by a better one if we get the chance. basically we decode the audio to signed linear 16bit samples for each packet we receive and iterate the samples and add up the absolute values then compute the average and compare it to the acceptable level considered "talking" then we use a counter to make sure it's in the same state several times in a row before actually changing the status of the channel to talking or not talking. A real vad algorithm would also be able to filter out the voice from the background noise and judge the energy based only on that. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: KMTDBOURSE <kmtdbourse(a)free.fr> To: freeswitch-dev(a)lists.freeswitch.org Sent: Thursday, February 14, 2008 1:51:55 PM Subject: Re: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, I ma interested in the VAD algorithm used where where is the code exactly? maybe a link any document support about the vad alogorithm used therein? thanks, Zelda At 01:47 14/02/2008, Anthony Minessale wrote: They are enabled by the vad params in the sofia profile xml <param name="vad"> Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Wednesday, February 13, 2008 10:17:23 AM Subject: Re: [Freeswitch-dev] VAD, TALK and NOTALK events Please correct me if I’m wrong. The events are indeed fired, but only if the SWITCH_VAD_FLAG_EVENTS_TALK or NOTALK flags are set for an rtp_session. I found no place in the code where those flags are set. That is what I meant by not being “enabled”. I subscribed to the events and received nothing. Then I set the flags on switch_rtp_enable_vad function, and then I started receiving the events. Perhaps there is another way of doing it that I’m missing. Cesar Cepeda. De: freeswitch-dev-bounces(a)lists.freeswitch.org [ mailto:freeswitch-dev-bounces@lists.freeswitch.org] En nombre de Anthony Minessale Enviado el: Miércoles, 13 de Febrero de 2008 08:17 Para: freeswitch-dev(a)lists.freeswitch.org Asunto: Re: [Freeswitch-dev] VAD, TALK and NOTALK events The events are fired. I don't know what you mean by "enabled" You can either bind an event to a callback in the load function of your custom module. or connect to mod_event_socket on tcp and request "events talk notalk" void my_event_handler(switch_event_t *event) { } if (switch_event_bind(modname, SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } if (switch_event_bind(modname, SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 12, 2008 6:53:03 PM Subject: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, >From looking at the code of switch_rtp.c I can see that VAD is done if enabled in the sip profile in order to not send packets when there is no noise. There are also two events {TALK, NOTALK} which can be fired when VAD kicks in, nevertheless these two events are enabled nowhere (or at least I didn't find a place in the code where they were enabled), so no matter if you ask for them you don't receive them. My questions are the following: Is there some particular reason for not enabling these two events? Is the event queue easily overflowed if these two events are enabled? Thanks. Cesar Cepeda. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org Never miss a thing. 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ATTENTION: Codec Module Changes applies to EVERYONE!
by Brian West 19 Feb '08

19 Feb '08

FreeSWITCHers, Today we removed mod_g711, mod_g722, mod_g726, mod_gsm and mod_l16, g711 and l16 were moved into the core but still use the loadable module interface because without these two you can't do anything. We now have mod_voipcodecs which covers G722, G726, GSM, IMA_ADPCM and LPC10. Changes would need to be made to your config files that are currently installed. 1. modules.conf in your source directory will need these changes: Remove: codecs/mod_l16 codecs/mod_gsm codecs/mod_g722 codecs/mod_g711 codecs/mod_g726 Add: codecs/mod_voipcodecs 2. modules.conf.xml in your prefix/conf/autoload_configs will need to be changed also: Remove: <load module="mod_l16"/> <load module="mod_gsm"/> <load module="mod_g722"/> <load module="mod_g711"/> <load module="mod_g726"/> Add: <load module="mod_voipcodecs"/> With this change we have gained working LPC10 @ 90ms and DVI4 (IMA_ADPCM) at 8k and 16k. Thanks, Brian West FreeSWITCH.org PS: please do not reply ALL to this email ;)

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Re: [Freeswitch-dev] VAD, TALK and NOTALK events
by Anthony Minessale 14 Feb '08

14 Feb '08

They are enabled by the vad params in the sofia profile xml <param name="vad"> Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Wednesday, February 13, 2008 10:17:23 AM Subject: Re: [Freeswitch-dev] VAD, TALK and NOTALK events  Please correct me if I’m wrong. The events are indeed fired, but only if the SWITCH_VAD_FLAG_EVENTS_TALK or NOTALK flags are set for an rtp_session. I found no place in the code where those flags are set. That is what I meant by not being “enabled”. I subscribed to the events and received nothing. Then I set the flags on switch_rtp_enable_vad function, and then I started receiving the events. Perhaps there is another way of doing it that I’m missing. Cesar Cepeda. De: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] En nombre de Anthony Minessale Enviado el: Miércoles, 13 de Febrero de 2008 08:17 Para: freeswitch-dev(a)lists.freeswitch.org Asunto: Re: [Freeswitch-dev] VAD, TALK and NOTALK events The events are fired. I don't know what you mean by "enabled" You can either bind an event to a callback in the load function of your custom module. or connect to mod_event_socket on tcp and request "events talk notalk" void my_event_handler(switch_event_t *event) { } if (switch_event_bind(modname, SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } if (switch_event_bind(modname, SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 12, 2008 6:53:03 PM Subject: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, >From looking at the code of switch_rtp.c I can see that VAD is done if enabled in the sip profile in order to not send packets when there is no noise. There are also two events {TALK, NOTALK} which can be fired when VAD kicks in, nevertheless these two events are enabled nowhere (or at least I didn't find a place in the code where they were enabled), so no matter if you ask for them you don't receive them. My questions are the following: Is there some particular reason for not enabling these two events? Is the event queue easily overflowed if these two events are enabled? Thanks. Cesar Cepeda. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org Never miss a thing. Make Yahoo your homepage. -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [Freeswitch-dev] VAD, TALK and NOTALK events
by Anthony Minessale 13 Feb '08

13 Feb '08

The events are fired. I don't know what you mean by "enabled" You can either bind an event to a callback in the load function of your custom module. or connect to mod_event_socket on tcp and request "events talk notalk" void my_event_handler(switch_event_t *event) { } if (switch_event_bind(modname, SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } if (switch_event_bind(modname, SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 12, 2008 6:53:03 PM Subject: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, >From looking at the code of switch_rtp.c I can see that VAD is done if enabled in the sip profile in order to not send packets when there is no noise. There are also two events {TALK, NOTALK} which can be fired when VAD kicks in, nevertheless these two events are enabled nowhere (or at least I didn't find a place in the code where they were enabled), so no matter if you ask for them you don't receive them. My questions are the following: Is there some particular reason for not enabling these two events? Is the event queue easily overflowed if these two events are enabled? Thanks. Cesar Cepeda. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-dev] [Freeswitch-users] Not able to make an outgoing Call through FXO....
by Irshad Ahamed 13 Feb '08

13 Feb '08

I am from India. The Indian dial tone is a combination of 350 Hz tone and a 440 Hz tone. Anyways here is the snippet of my tones.conf @ /etc/openzap/ [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,480,620) detect-busy => 480,620 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 BR, Irshad. On Jan 10, 2008 6:57 AM, Michael Jerris <mike(a)jerris.com> wrote: > Yes, these are set in tones.conf. You will need to add the correct values > for your area (we can add them to the default config if you can provide > those). Also, please update to the current code, tony just added quite a > few features to mod_openzap including support for hold music, transfer > (blind and attended), and 3-way. > Mike > > > On Jan 9, 2008, at 7:57 AM, Irshad Ahamed wrote: > > With the same set-up, Outgoing calls are working fine with Asterisk. > > Are there any tools to detect the presence of dial tone??? If it's any > parameter issue, can we adjust those parameters??? > > NOTE: I copied tones.conf available from the source code to /etc/openzap/ > > BR, > Irshad. > > > On Jan 5, 2008 12:00 AM, Anthony Minessale <anthmct(a)yahoo.com > wrote: > > > I think it's waiting for a dialtone before it sends the digits > > and then giving up because it can't fine one. > > > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Irshad Ahamed < irshad.fswitch(a)gmail.com> > > To: freeswitch-users(a)lists.freeswitch.org; > > freeswitch-dev(a)lists.freeswitch.org > > Sent: Friday, January 4, 2008 4:24:07 AM > > Subject: Re: [Freeswitch-users] Not able to make an outgoing Call > > through FXO.... > > > > Hi Anthony, > > > > Thanks for the info. By adding the span-ID = 2 , I was able to receive > > an incoming call successfully. > > The dialplan/default.xml is as below: > > <extension name="incoming"> > > <condition field="destination_number" expression="^(1)$"> > > <action application="bridge" data="openzap/1/1"/> > >  > > </condition> > > </extension> > > > > <extension name="outgoing-fxo"> > > <condition field="destination_number" expression="^(43)$"> > > <action application="bridge" data="openzap/2/1/43"/> > > </condition> > > </extension> > > > > The Openzap.conf.xml looks like below: > > > > <configuration name="openzap.conf" description="OpenZAP Configuration"> > > <settings> > > <param name="debug" value="9"/> > > </settings> > > <analog_spans> > > > > <param name="tonegroup" value="us"/> > > <param name="digit-timeout" value="2000"/> > > <param name="max-digits" value="11"/> > > <param name="dialplan" value="XML"/> > > <param name="context" value="default"/> > >  > >  > >  > >  > > > > > > <param name="tonegroup" value="us"/> > > <param name="digit-timeout" value="2000"/> > > <param name="max-digits" value="1"/> > > <param name="dialplan" value="XML"/> > > <param name="context" value="default"/> > >  > >  > >  > >  > > > > </analog_spans> > > </configuration> > > > > But still I am unable to make an Outgoing Call. Please have a look at > > the Freeswitch Log below: > > > > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:664 process_event() EVENT > > [OFFHOOK][1:1] STATE [DOWN] > > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:726 process_event() Changing > > state on 1:1 from DOWN to DIALTONE > > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:184 zap_analog_channel_run() > > ANALOG CHANNEL thread starting. > > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for DIALTONE > > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:490 zap_analog_channel_run() > > DTMF 4 > > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:492 zap_analog_channel_run() > > Changing state on 1:1 from DIALTONE to COLLECT > > 2008-01-04 15:45:07 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > sig [COLLECTED_DIGIT] > > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for COLLECT > > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:490 zap_analog_channel_run() > > DTMF 2 > > 2008-01-04 15:45:07 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > sig [COLLECTED_DIGIT] > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:507 zap_analog_channel_run() > > Number obtained [42] > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:508 zap_analog_channel_run() > > Changing state on 1:1 from COLLECT to IDLE > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for IDLE > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > sig [START] > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU > > 20ms > > 2008-01-04 15:45:09 [NOTICE] switch_channel.c:392 > > switch_channel_set_name() New Chan OpenZAP/42 > > [8d46eb7f-4305-49e8-a4a5-eb97af3b696b] > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:892 zap_channel_from_event() > > OpenZAP/42 State Change CS_NEW -> CS_INIT > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:397 > > switch_core_session_run() (OpenZAP/42) State INIT > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:256 channel_on_init() > > OpenZAP/42 State Change CS_INIT -> CS_RING > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:400 > > switch_core_session_run() (OpenZAP/42) State RING > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:276 channel_on_ring() > > OpenZAP/42 CHANNEL RING > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:64 > > switch_core_standard_on_ring() Standard RING OpenZAP/42 > > 2008-01-04 15:45:09 [INFO] mod_dialplan_xml.c:217 dialplan_hunt() > > Processing OpenZAP->42! > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > > conditions ${numbering_plan}() =~ // > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > > conditions destination_number(42) =~ /^(1)$/ > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:108 parse_exten() Regex > > mismatch > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > > conditions destination_number(42) =~ /^(43)$/ > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:108 parse_exten() Regex > > mismatch > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > > conditions destination_number(42) =~ /^(42)$/ > > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:273 dialplan_hunt() > > OpenZAP/42 State Change CS_RING -> CS_EXECUTE > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:406 > > switch_core_session_run() (OpenZAP/42) State EXECUTE > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:294 channel_on_execute() > > OpenZAP/42 CHANNEL EXECUTE > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:121 > > switch_core_standard_on_execute() Standard EXECUTE > > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute > > set(use_profile=${cond(${is_lan_addr(${network_addr})} == yes ? nat : > > default)}) > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:158 > > switch_core_standard_on_execute() Expanded String set(use_profile=default) > > 2008-01-04 15:45:09 [DEBUG] mod_dptools.c:419 set_function() SET > > [use_profile]=[default] > > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute set_user(default@${domain}) > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:158 > > switch_core_standard_on_execute() Expanded String set_user( > > default(a)192.168.2.51) > > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute info() > > 2008-01-04 15:45:09 [CRIT] mod_dptools.c:630 info_function() > > CHANNEL_DATA: > > Channel-State: CS_EXECUTE > > Channel-State-Number: 4 > > Channel-Name: OpenZAP/42 > > Unique-ID: 8d46eb7f-4305-49e8-a4a5-eb97af3b696b > > Channel-Read-Codec-Name: PCMU > > Channel-Read-Codec-Rate: 8000 > > Channel-Write-Codec-Name: PCMU > > Channel-Write-Codec-Rate: 8000 > > Caller-Username: OpenZAP > > Caller-Dialplan: XML > > Caller-Caller-ID-Name: OpenZAP > > Caller-Caller-ID-Number: 1 > > Caller-Destination-Number: 42 > > Caller-Unique-ID: 8d46eb7f-4305-49e8-a4a5-eb97af3b696b > > Caller-Source: mod_openzap > > Caller-Context: default > > Caller-Channel-Name: OpenZAP/42 > > Caller-Channel-Created-Time: 1199441709953187 > > Caller-Channel-Answered-Time: 0 > > Caller-Channel-Hangup-Time: 0 > > Caller-Channel-Transfer-Time: 0 > > Caller-Screen-Bit: yes > > Caller-Privacy-Hide-Name: no > > Caller-Privacy-Hide-Number: no > > variable_use_profile: default > > variable_numbering_plan: US > > variable_user_name: default > > variable_domain_name: 192.168.2.51 > > > > > > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute answer() > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:663 > > channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:339 > > switch_core_session_receive_message() Kill OpenZAP/42 [BREAK] > > 2008-01-04 15:45:09 [NOTICE] mod_dptools.c:361 answer_function() Channel > > [OpenZAP/42] has been answered > > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute bridge(openzap/2/1/42)2008-01-04 > > 15:45:09 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU 20ms > > 2008-01-04 15:45:09 [NOTICE] switch_channel.c:392 > > switch_channel_set_name() New Chan OPENZAP/42 > > [bef803ed-e932-4a22-84e7-9131070fe771] > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:802 channel_outgoing_channel() > > OPENZAP/42 State Change CS_NEW -> CS_INIT > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for UP > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > sig [UP] > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:55 analog_fxo_outgoing_call() > > Changing state on 2:1 from DOWN to DIALING > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:184 zap_analog_channel_run() > > ANALOG CHANNEL thread starting. > > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 2:1 for DIALING > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:397 > > switch_core_session_run() (OPENZAP/42) State INIT > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:256 channel_on_init() > > OPENZAP/42 State Change CS_INIT -> CS_RING > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > > 2008-01-04 15:45:09 [DEBUG] switch_ivr_originate.c:625 > > switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel > > 20ms > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:400 > > switch_core_session_run() (OPENZAP/42) State RING > > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:276 channel_on_ring() > > OPENZAP/42 CHANNEL RING > > 2008-01-04 15:45:09 [DEBUG] switch_ivr_originate.c:48 > > originate_on_ring() OPENZAP/42 State Change CS_RING -> CS_HOLD > > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:415 > > switch_core_session_run() (OPENZAP/42) State HOLD > > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:204 > > switch_core_standard_on_hold() Standard HOLD > > > > > > freeswitch(a)sipxpbx.oneconv.com> > > > > freeswitch(a)sipxpbx.oneconv.com > > > > > freeswitch(a)sipxpbx.oneconv.com> 2008-01-04 15:45:39 [DEBUG] > > zap_analog.c:244 zap_analog_channel_run() Changing state on 2:1 from DIALING > > to BUSY > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 2:1 for BUSY > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:467 zap_analog_channel_run() > > Changing state on 2:1 from BUSY to DOWN > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 2:1 for DOWN > > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:916 on_fxo_signal() got FXO > > sig 2:1 [STOP] > > 2008-01-04 15:45:40 [NOTICE] mod_openzap.c:934 on_fxo_signal() Hangup > > OPENZAP/42 [CS_HOLD] [NORMAL_CIRCUIT_CONGESTION] > > 2008-01-04 15:45:40 [DEBUG] switch_channel.c:1071 > > switch_channel_perform_hangup() Kill OPENZAP/42 [KILL] > > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:648 zap_analog_channel_run() > > ANALOG CHANNEL thread ended. > > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:394 > > switch_core_session_run() (OPENZAP/42) State HANGUP > > 2008-01-04 15:45:40 [WARNING] mod_openzap.c:319 channel_on_hangup() VETO > > Changing state on 2:1 from DOWN to HANGUP > > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:359 channel_on_hangup() > > OPENZAP/42 CHANNEL HANGUP > > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:45 > > switch_core_standard_on_hangup() Standard HANGUP OPENZAP/42, cause: > > NORMAL_CIRCUIT_CONGESTION > > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:683 > > switch_core_session_thread() Session 4 (OPENZAP/42) Locked, Waiting on > > external entities > > 2008-01-04 15:45:40 [DEBUG] switch_ivr_originate.c:873 > > switch_ivr_originate() Originate Resulted in Error Cause: 34 > > [NORMAL_CIRCUIT_CONGESTION] > > 2008-01-04 15:45:40 [INFO] mod_dptools.c:1331 audio_bridge_function() > > Originate Failed. Cause: NORMAL_CIRCUIT_CONGESTION > > 2008-01-04 15:45:40 [NOTICE] mod_dptools.c:1358 audio_bridge_function() > > Hangup OpenZAP/42 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] > > 2008-01-04 15:45:40 [DEBUG] switch_channel.c:1071 > > switch_channel_perform_hangup() Kill OpenZAP/42 [KILL] > > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:394 > > switch_core_session_run() (OpenZAP/42) State HANGUP > > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:329 channel_on_hangup() > > Changing state on 1:1 from UP to HANGUP > > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:359 channel_on_hangup() > > OpenZAP/42 CHANNEL HANGUP > > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:45 > > switch_core_standard_on_hangup() Standard HANGUP OpenZAP/42, cause: > > NORMAL_CIRCUIT_CONGESTION > > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:683 > > switch_core_session_thread() Session 3 (OpenZAP/42) Locked, Waiting on > > external entities > > 2008-01-04 15:45:40 [INFO] switch_core_session.c:689 > > switch_core_session_thread() Session 3 (OpenZAP/42) Ended > > 2008-01-04 15:45:40 [NOTICE] switch_core_session.c:691 > > switch_core_session_thread() Close Channel OpenZAP/42 [CS_HANGUP] > > 2008-01-04 15:45:40 [INFO] switch_core_session.c:689 > > switch_core_session_thread() Session 4 (OPENZAP/42) Ended > > 2008-01-04 15:45:40 [NOTICE] switch_core_session.c:691 > > switch_core_session_thread() Close Channel OPENZAP/42 [CS_HANGUP] > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for HANGUP > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:290 zap_analog_channel_run() > > Changing state on 1:1 from HANGUP to BUSY > > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for BUSY > > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:664 process_event() EVENT > > [ONHOOK][1:1] STATE [BUSY] > > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:690 process_event() Changing > > state on 1:1 from BUSY to DOWN > > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for DOWN > > 2008-01-04 15:45:53 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > sig [STOP] > > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:648 zap_analog_channel_run() > > ANALOG CHANNEL thread ended. > > > > Thanks, > > Irshad. > > > > > > > > > > > > On Jan 3, 2008 7:43 PM, Anthony Minessale < anthmct(a)yahoo.com> wrote: > > > > > What does openzap.conf.xml look like? > > > Perhaps you are using the default which only has 1 span defined. > > > Try cloning the whole block so there > > > is a span 1 and 2 like this: > > > > > > <configuration name=" openzap.conf" description="OpenZAP > > > Configuration"> > > > <settings> > > > <param name="debug" value="0"/> > > > </settings> > > > <analog_spans> > > > > > > <param name="tonegroup" value="us"/> > > > <param name="digit-timeout" value="2000"/> > > > <param name="max-digits" value="11"/> > > > <param name="dialplan" value="XML"/> > > > <param name="context" value="default"/> > > >  > > >  > > >  > > >  > > > > > > > > > > > > <param name="tonegroup" value="us"/> > > > <param name="digit-timeout" value="2000"/> > > > <param name="max-digits" value="11"/> > > > <param name="dialplan" value="XML"/> > > > <param name="context" value="default"/> > > >  > > >  > > >  > > >  > > > > > > > > > > > > </analog_spans> > > > </configuration> > > > > > > > > > > > > > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale@hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888@conference.freeswitch.org > > > iax:guest@conference.freeswitch.org/888 > > > googletalk:conf+888@conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > > > > ----- Original Message ---- > > > From: Irshad Ahamed <irshad.fswitch(a)gmail.com > > > > To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org > > > > > > Sent: Wednesday, January 2, 2008 12:40:14 AM > > > Subject: [Freeswitch-users] Not able to make an outgoing Call through > > > FXO.... > > > > > > Hi, > > > > > > Wishing U all a very Happy n Prosperous New Year!!!! > > > > > > I am trying to make an outgoing call using FXO connected to an > > > external telephone line. > > > > > > Here is the set-up: > > > > > > 1. FXO connected to external telephone line > > > 2. FXS connected to analog phone. > > > > > > > > > The Openzap.conf @ /etc/openzap is as follows: > > > [span zt] > > > name => OpenZAP > > > number => 1 > > > fxs-channel => 1 > > > > > > [span zt] > > > name => OpenZAP > > > number => 1 > > > fxo-channel => 4 > > > > > > The default.xml @ /usr/local/freeswitch/conf/dialplan is configured > > > for inward and outgoing calls on FXO as follows: > > > > > > <extension name="outgoing-fxo"> > > > <condition field="destination_number" expression="^(43)$"> > > > <action application="bridge" data="openzap/2/1/43"/> > > > </condition> > > > </extension> > > > > > > <extension name="incoming-fxo"> > > > <condition field="destination_number" expression="^(35)$"> > > > <action application="bridge" data="openzap/1/1"/> > > > </condition> > > > </extension> > > > > > > While Start-up Freeswitch following logs were seen: > > > > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1770 load_config() found config > > > for span > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1797 load_config() created span 1 > > > of type zt > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 > > > [name]=[OpenZAP] > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 > > > [number]=[1] > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 > > > [fxs-channel]=[1] > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1850 load_config() setting trunk > > > type to 'FXS' start(KEWL) > > > 2008-01-02 11:51:22 [INFO] zap_zt.c:168 zt_open_range() configuring > > > device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:31 > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1770 load_config() found config > > > for span > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1797 load_config() created span 2 > > > of type zt > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 > > > [name]=[OpenZAP] > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 > > > [number]=[1] > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 > > > [fxo-channel]=[4] > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1839 load_config() setting trunk > > > type to 'FXO' start(KEWL) > > > 2008-01-02 11:51:22 [INFO] zap_zt.c:168 zt_open_range() configuring > > > device /dev/zap/channel channel 4 as OpenZAP device 2:1 fd:32 > > > 2008-01-02 11:51:22 [INFO] zap_io.c:1881 load_config() Configured 2 > > > channel(s) > > > 2008-01-02 11:51:22 [CONSOLE] switch_loadable_module.c:748 > > > switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > > > > > > and while I made a call following logs were seen: > > > > > > 2008-01-02 12:04:16 [NOTICE] switch_core_state_machine.c:134 > > > switch_core_standard_on_execute() Execute bridge(openzap/2/1/43)2008-01-02 > > > 12:04:16 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU 20ms > > > 2008-01-02 12:04:16 [NOTICE] switch_channel.c:392 > > > switch_channel_set_name() New Chan OPENZAP/43 > > > [e3630387-25d6-4031-bf81-239c3e64e3fa] > > > 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:802 > > > channel_outgoing_channel() OPENZAP/43 State Change CS_NEW -> CS_INIT > > > 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:625 > > > switch_core_session_signal_state_change() Kill OPENZAP/43 [BREAK] > > > 2008-01-02 12:04:16 [ERR] zap_io.c:884 zap_channel_outgoing_call() > > > outgoing_call method not implemented! > > > 2008-01-02 12:04:16 [NOTICE] mod_openzap.c:818 > > > channel_outgoing_channel() Close Channel OPENZAP/43 [CS_INIT] > > > 2008-01-02 12:04:16 [ERR] switch_ivr_originate.c:525 > > > switch_ivr_originate() Cannot Create Outgoing Channel! cause: > > > DESTINATION_OUT_OF_ORDER > > > 2008-01-02 12:04:16 [DEBUG] switch_ivr_originate.c:873 > > > switch_ivr_originate() Originate Resulted in Error Cause: 27 > > > [DESTINATION_OUT_OF_ORDER] > > > 2008-01-02 12:04:16 [INFO] mod_dptools.c:1331 audio_bridge_function() > > > Originate Failed. Cause: DESTINATION_OUT_OF_ORDER > > > 2008-01-02 12:04:16 [NOTICE] mod_dptools.c:1358 > > > audio_bridge_function() Hangup OpenZAP/43 [CS_EXECUTE] > > > [DESTINATION_OUT_OF_ORDER] > > > 2008-01-02 12:04:16 [DEBUG] switch_channel.c:1071 > > > switch_channel_perform_hangup() Kill OpenZAP/43 [KILL] > > > 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:625 > > > switch_core_session_signal_state_change() Kill OpenZAP/43 [BREAK] > > > 2008-01-02 12:04:16 [DEBUG] switch_core_state_machine.c:394 > > > switch_core_session_run() (OpenZAP/43) State HANGUP > > > 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:329 channel_on_hangup() > > > Changing state on 1:1 from IDLE to HANGUP > > > 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:359 channel_on_hangup() > > > OpenZAP/43 CHANNEL HANGUP > > > 2008-01-02 12:04:16 [DEBUG] switch_core_state_machine.c:45 > > > switch_core_standard_on_hangup() Standard HANGUP OpenZAP/43, cause: > > > DESTINATION_OUT_OF_ORDER > > > 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:683 > > > switch_core_session_thread() Session 1 (OpenZAP/43) Locked, Waiting on > > > external entities > > > 2008-01-02 12:04:16 [INFO] switch_core_session.c:689 > > > switch_core_session_thread() Session 1 (OpenZAP/43) Ended > > > 2008-01-02 12:04:16 [NOTICE] switch_core_session.c:691 > > > switch_core_session_thread() Close Channel OpenZAP/43 [CS_HANGUP] > > > 2008-01-02 12:04:16 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > > Executing state handler on 1:1 for HANGUP > > > 2008-01-02 12:04:17 [DEBUG] zap_analog.c:290 zap_analog_channel_run() > > > Changing state on 1:1 from HANGUP to BUSY > > > 2008-01-02 12:04:17 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > > Executing state handler on 1:1 for BUSY > > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:664 process_event() EVENT > > > [ONHOOK][1:1] STATE [BUSY] > > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:690 process_event() Changing > > > state on 1:1 from BUSY to DOWN > > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > > Executing state handler on 1:1 for DOWN > > > 2008-01-02 12:04:20 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > > sig [STOP] > > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:648 zap_analog_channel_run() > > > ANALOG CHANNEL thread ended. > > > > > > Could anyone help me out in getting FXO working ??? > > > > > > Thanks in advance... > > > > > > BR, > > > Irshad. > > > > > > > > > > > > ------------------------------ > > > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try > > > it now.<http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8…> > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > ------------------------------ > > Looking for last minute shopping deals? Find them fast with Yahoo! > > Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

1 1

Hey guys
by Brian West 13 Feb '08

13 Feb '08

Check it out http://www.frappr.com/freeswitchusers Don't forget to add yourself. /b

1 0

Prevent call setup
by Alois Komenda 13 Feb '08

13 Feb '08

Hello, how can I prevent the completion of the call setup if it was detected (by my module or a client of the event_socket server for example) that the call is not allowed? The criteria for a call to be not allowed might be out of the scope of the dialplan so I can't use dialplan logic for this task. What happens if I send an event like CHANNEL_HANGUP or CHANNEL_DESTROY from an event_socket client? Does this initiate any actions? Thanks in advance Regards, -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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Re: [Freeswitch-dev] TDM questions: OZ, signaling on digital spans
by Anthony Minessale 06 Feb '08

06 Feb '08

see inline comments. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Collins <mcollins(a)fcnetwork.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 5, 2008 6:34:50 PM Subject: Re: [Freeswitch-dev] TDM questions: OZ, signaling on digital spans  From: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, February 05, 2008 6:19 AM To: freeswitch-dev(a)lists.freeswitch.org Subject: Re: [Freeswitch-dev] TDM questions: OZ, signaling on digital spans It's a trickle-up methodology. The low level driver interprets the voltage change or the bits flipping or whatever and turns it into driver specific events which in turn are changed into openzap specific events by the zap_<whatver>.c zap_zt.c, zap_pika.c etc. when you apply a signaling to a span in the mod_openzap config the signal specific module will monitor the span for these events and decide what they mean in reference to whatever signaling type it represents. Okay, so the driver-specific events get captured and translated… Makes sense. Curious: does this all happen at OSI layer 2? I don't know exactly how it works. so voltage changes on zaptel card zaptel specific ring event is caught by zap_zt.c openzap ring event is queued on the channel and event data flag is set on the span. zap_analog.c sees the event data flag and gets the ring event on the channel ring event is sent up to mod_openzap who starts a channel in freeswitch. so implementing another sig type like e&m just requires zap_em.c and proper interpretation of the bit shifting events. I belive zaptel drivers and pika can do it but I don't know if the native sangoma can yet. I think I get it. Next question: on a digital span that is set up in the <analog spans> section of openzap.conf.xml – does zap_analog.c do the signal translation even though the span is technically digital? It almost sounds like zap_analog.c is agnostic to the analog/digital nature of the span simply because it’s receiving events from the driver via openzap… Last question: if zap_analog.c is handling the translation of FXO/FXS signaling then couldn’t we almost just copy zap_analog.c to zap_em.c and then redefine the bit patterns that represent on-hook, ringing, hangup, etc.? Or am I oversimplifying again? ;) Yes that is more or less correct. In fact once you took out all the analog state machine etc from zap_analog.c zap_em.c will be rather tiny just starting calls when it gets off hook and ending them when it gets onhook. Thanks for the explanation – I will distill this thread into it’s components and add it to the wiki when we’re done. -MC P.S. – the zaptel drivers do support E&M, E&M wink, etc. I’ve used E&M with the orange splat and connected to my NEC PBX with the exact same tor2 and trunk-side T1 card. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Collins <mcollins(a)fcnetwork.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 5, 2008 1:39:13 AM Subject: [Freeswitch-dev] TDM questions: OZ, signaling on digital spans Guys, You better run for cover because Mikey has been reading technical books again and now he has questions! J First, I was reading in Matthew Gast’s T1 – A Survival Guide about channelized T1’s and channel associated signaling. Just to make sure I am understanding it correctly – CAS and robbed bit essentially refer to the same thing on a channelized T1 don’t they? The bits that are robbed include the signaling bits – AB or ABCD depending on whether you have SF or ESF. Please confirm if I am understanding this correctly. Second, aren’t FXO and FXS signaling on a digital channel really just a matter of interpreting the meanings of the A and B bits being high or low? Obviously there isn’t an actual loop start or ground start on a digital channel, but rather just a nomenclature for deciding that a particular combination of A and B bits being high and/or low has a meaning, like A high + B low means on-hook in one signaling type but could mean ringing in another signaling type. Again, correct me if I’m wrong… Third, if FXO/FXS signaling is just a matter of manipulating the AB bits in CAS, then shouldn’t it be relatively easy to add E&M signaling since that is essentially just another set of meanings for each AB bit pattern? Or am I oversimplifying it? Last question – how does OZ, in zap_zt.c, determine the signal start type? (E.g. see code starting at line 87.) AFAIK it is not set in openzap.conf.xml nor in openzap.conf. Does it get that info from the zaptel driver? I know the start type is set in zaptel.conf, I just didn’t know if the OZ was smart enough to figure out the start type without requiring the user to specify it in a conf file. Thanks for letting me ask all these questions! I’ll do my best to update the wiki as appropriate as I get the answers… -MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [Freeswitch-dev] TDM questions: OZ, signaling on digital spans
by Anthony Minessale 06 Feb '08

06 Feb '08

It's a trickle-up methodology. The low level driver interprets the voltage change or the bits flipping or whatever and turns it into driver specific events which in turn are changed into openzap specific events by the zap_<whatver>.c zap_zt.c, zap_pika.c etc. when you apply a signaling to a span in the mod_openzap config the signal specific module will monitor the span for these events and decide what they mean in reference to whatever signaling type it represents. so voltage changes on zaptel card zaptel specific ring event is caught by zap_zt.c openzap ring event is queued on the channel and event data flag is set on the span. zap_analog.c sees the event data flag and gets the ring event on the channel ring event is sent up to mod_openzap who starts a channel in freeswitch. so implementing another sig type like e&m just requires zap_em.c and proper interpretation of the bit shifting events. I belive zaptel drivers and pika can do it but I don't know if the native sangoma can yet. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Collins <mcollins(a)fcnetwork.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 5, 2008 1:39:13 AM Subject: [Freeswitch-dev] TDM questions: OZ, signaling on digital spans  Guys, You better run for cover because Mikey has been reading technical books again and now he has questions! J First, I was reading in Matthew Gast’s T1 – A Survival Guide about channelized T1’s and channel associated signaling. Just to make sure I am understanding it correctly – CAS and robbed bit essentially refer to the same thing on a channelized T1 don’t they? The bits that are robbed include the signaling bits – AB or ABCD depending on whether you have SF or ESF. Please confirm if I am understanding this correctly. Second, aren’t FXO and FXS signaling on a digital channel really just a matter of interpreting the meanings of the A and B bits being high or low? Obviously there isn’t an actual loop start or ground start on a digital channel, but rather just a nomenclature for deciding that a particular combination of A and B bits being high and/or low has a meaning, like A high + B low means on-hook in one signaling type but could mean ringing in another signaling type. Again, correct me if I’m wrong… Third, if FXO/FXS signaling is just a matter of manipulating the AB bits in CAS, then shouldn’t it be relatively easy to add E&M signaling since that is essentially just another set of meanings for each AB bit pattern? Or am I oversimplifying it? Last question – how does OZ, in zap_zt.c, determine the signal start type? (E.g. see code starting at line 87.) AFAIK it is not set in openzap.conf.xml nor in openzap.conf. Does it get that info from the zaptel driver? I know the start type is set in zaptel.conf, I just didn’t know if the OZ was smart enough to figure out the start type without requiring the user to specify it in a conf file. Thanks for letting me ask all these questions! I’ll do my best to update the wiki as appropriate as I get the answers… -MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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TDM questions: OZ, signaling on digital spans
by Michael Collins 05 Feb '08

05 Feb '08

Guys, You better run for cover because Mikey has been reading technical books again and now he has questions! :-) First, I was reading in Matthew Gast's T1 - A Survival Guide about channelized T1's and channel associated signaling. Just to make sure I am understanding it correctly - CAS and robbed bit essentially refer to the same thing on a channelized T1 don't they? The bits that are robbed include the signaling bits - AB or ABCD depending on whether you have SF or ESF. Please confirm if I am understanding this correctly. Second, aren't FXO and FXS signaling on a digital channel really just a matter of interpreting the meanings of the A and B bits being high or low? Obviously there isn't an actual loop start or ground start on a digital channel, but rather just a nomenclature for deciding that a particular combination of A and B bits being high and/or low has a meaning, like A high + B low means on-hook in one signaling type but could mean ringing in another signaling type. Again, correct me if I'm wrong... Third, if FXO/FXS signaling is just a matter of manipulating the AB bits in CAS, then shouldn't it be relatively easy to add E&M signaling since that is essentially just another set of meanings for each AB bit pattern? Or am I oversimplifying it? Last question - how does OZ, in zap_zt.c, determine the signal start type? (E.g. see code starting at line 87.) AFAIK it is not set in openzap.conf.xml nor in openzap.conf. Does it get that info from the zaptel driver? I know the start type is set in zaptel.conf, I just didn't know if the OZ was smart enough to figure out the start type without requiring the user to specify it in a conf file. Thanks for letting me ask all these questions! I'll do my best to update the wiki as appropriate as I get the answers... -MC

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