January 2008 - freeswitch-dev@lists.freeswitch.org

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

Re: [Freeswitch-dev] SIP Over TCP/TLS
by Imthiyaz Ahmed 31 Jan '08

31 Jan '08

Thanks Brian I am going to use sangoma modules for PSTN connectivity. I think I don't need asterisk in this scenario. Microsoft Mediation server <---------->FreeSwitch with Sangoma Modules installed. SIP over TCP/TLS <--------------->TDM That works right ? Thanks Imthiyaz

4 10

[Freeswitch-dev] Ultrawideband
by Brian West 31 Jan '08

31 Jan '08

Anyone know of any devices and what codecs currently support ultra- wideband? Btw portaudio@32k via FS@32k to FS@32k works. /b

2 4

[Freeswitch-dev] LPC10 Codec
by Brian West 30 Jan '08

30 Jan '08

Does anyone really find any use for this codec? It's currently broken in tree and wondered if it was worth the time fixing? Chime in. /b

3 4

[Freeswitch-dev] SIP Over TCP/TLS
by Imthiyaz Ahmed 28 Jan '08

28 Jan '08

---------- Forwarded message ---------- From: Imthiyaz Ahmed <imthiyazg(a)gmail.com> Date: Jan 28, 2008 7:07 PM Subject: SIP Over TCP/TLS To: freeswitch-dev(a)lists.freeswitch.org. Hello Does Free switch supports SIP Over TCP/TLS ? I am thinking of building a Media Gateway for OCS 2007 . Thanks Imthiyaz

2 1

Re: [Freeswitch-dev] Call Hold ans swapping on FXS
by Anthony Minessale 28 Jan '08

28 Jan '08

I changed it to be options you have to set.    in <settings> for global or in for local *note* you cannot have both of these analog-options at once because they compete. 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: sanjeev mudholkar <sanjeev.mudholkar(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, January 25, 2008 1:18:29 AM Subject: [Freeswitch-dev] Call Hold ans swapping on FXS Hi, I'am trying to use Freeswitch(with FXS) as a UA. Everything is fine for a outgoing and incoming call. 1)On doing a hold (flash) on the FXS the call is put on hold but there no SIP signaling (i.e re-invite is not sent to the peer). How can I force freeswitch to send a re-invite? 2) I'am not able to establish a consultation call as it is conferenced with the first one as soon as the second call is connected. This also prevents from swaping between calls. The call swapping feature seems to have been removed by introducing the transfer/conference feature. Is there away to achieve call swapping ? Appreciate your help. Thanks in advance -Sanjeev ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

3 2

Re: [Freeswitch-dev] Call Hold ans swapping on FXS
by Anthony Minessale 25 Jan '08

25 Jan '08

We do not send a reinvite when you put a sip call on hold. if you have the hold-music param set in the settings section or in each span you can deliver moh to the on-hold party. The call swapping concept and the 3-way feature are competing features so we would need to design config options to expose them. Judging from your numerous feature requests on TDM you should consider sponsoring the OpenZAP project because our resources are limited and TDM users such as yourself should appreciate the goals of a BSD-licensed unified TDM library. Everything you see so far in OpenZAP has been done out of faith that sponsors will step up and try to steer the project towards it's eventual goals of bigger and better ISDN support/ss7 etc. 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: sanjeev mudholkar <sanjeev.mudholkar(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, January 25, 2008 1:18:29 AM Subject: [Freeswitch-dev] Call Hold ans swapping on FXS Hi, I'am trying to use Freeswitch(with FXS) as a UA. Everything is fine for a outgoing and incoming call. 1)On doing a hold (flash) on the FXS the call is put on hold but there no SIP signaling (i.e re-invite is not sent to the peer). How can I force freeswitch to send a re-invite? 2) I'am not able to establish a consultation call as it is conferenced with the first one as soon as the second call is connected. This also prevents from swaping between calls. The call swapping feature seems to have been removed by introducing the transfer/conference feature. Is there away to achieve call swapping ? Appreciate your help. Thanks in advance -Sanjeev ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

1 0

[Freeswitch-dev] Call Waiting not working!!!!!
by Irshad Ahamed 25 Jan '08

25 Jan '08

Hi, I am working on Freeswitch@ rev 7309. I remember with earlier revision call waiting was working fine. But now on getting the second incoming call FXS is returning 502 (BAD gateway error). Was there any change on FXS side pertaining to number of calls allowed or something??? Freeswitch Log: 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(dialed_ext=6601) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [dialed_ext]=[6601] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(hangup_after_bridge=true) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [hangup_after_bridge]=[true] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(bypass_media=false) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [bypass_media]=[false] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute bridge(openzap/1/1) 2008-01-25 15:48:17 [ERR] mod_openzap.c:833 channel_outgoing_channel() No channels available 2008-01-25 15:48:17 [ERR] switch_ivr_originate.c:827 switch_ivr_originate() Cannot Create Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_ivr_originate.c:1226 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [INFO] mod_dptools.c:1325 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [NOTICE] mod_dptools.c:1352 audio_bridge_function() Hangup sofia/default/6502@192.168.2.51:5060 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/6502@192.168.2.51:5060 [KILL] 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/6502@192.168.2.51:5060 [BREAK] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/6502@192.168.2.51:5060 Running State Change CS_HANGUP 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/6502@192.168.2.51:5060) State HANGUP 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/6502@192.168.2.51:5060 hanging up, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:332 sofia_on_hangup() Responding to INVITE with: 502 2008-01-25 15:48:17 [DEBUG] sofia.c:183 sofia_event_callback() event [nua_i_state] status [502][Call state] session: sofia/default/6502@192.168.2.51:5060 2008-01-25 15:48:17 [DEBUG] sofia.c:1430 sofia_handle_sip_i_state() Channel sofia/default/6502@192.168.2.51:5060 entering state [terminated] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/6502@192.168.2.51:5060, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Locked, Waiting on external entities 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Ended 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/6502@192.168.2.51:5060 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:665 process_event() EVENT [ONHOOK][1:1] STATE [UP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:691 process_event() Changing state on 1:1 from UP to DOWN 2008-01-25 15:48:36 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:1037 on_fxs_signal() got FXS sig [STOP] 2008-01-25 15:48:36 [NOTICE] mod_openzap.c:1115 on_fxs_signal() Hangup OpenZAP/777 [CS_EXECUTE] [NORMAL_CLEARING] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill OpenZAP/777 [KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:649 zap_analog_channel_run() ANALOG CHANNEL thread ended. 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:186 audio_bridge_thread() read: OpenZAP/777 Bad Frame.... Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/777] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:693 switch_ivr_multi_threaded_bridge() sofia/default/777(a)sipx.oneconv.com State Change CS_LOOPBACK -> CS_RESET 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() OpenZAP/777 Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/777) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:418 channel_on_hangup() OpenZAP/777 CHANNEL HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP OpenZAP/777, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 1 (OpenZAP/777) Locked, Waiting on external entities 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:180 audio_bridge_thread() write: OpenZAP/777 Bad Frame....[160] Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/777(a)sipx.oneconv.com] 2008-01-25 15:48:36 [NOTICE] switch_ivr_bridge.c:257 audio_bridge_on_loopback() Hangup sofia/default/777(a)sipx.oneconv.com[CS_RESET] [NORMAL_CLEARING] 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 1 (OpenZAP/777) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel OpenZAP/777 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/777(a)sipx.oneconv.com[KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/777(a)sipx.oneconv.com Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/777(a)sipx.oneconv.com) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/777(a)sipx.oneconv.com hanging up, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:325 sofia_on_hangup() Sending BYE to sofia/default/777(a)sipx.oneconv.com 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/777(a)sipx.oneconv.com, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Locked, Waiting on external entities 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/777(a)sipx.oneconv.com [CS_HANGUP] Thanks in advance... BR, Irshad.

1 1

[Freeswitch-dev] Call Hold ans swapping on FXS
by sanjeev mudholkar 25 Jan '08

25 Jan '08

Hi, I'am trying to use Freeswitch(with FXS) as a UA. Everything is fine for a outgoing and incoming call. 1)On doing a hold (flash) on the FXS the call is put on hold but there no SIP signaling (i.e re-invite is not sent to the peer). How can I force freeswitch to send a re-invite? 2) I'am not able to establish a consultation call as it is conferenced with the first one as soon as the second call is connected. This also prevents from swaping between calls. The call swapping feature seems to have been removed by introducing the transfer/conference feature. Is there away to achieve call swapping ? Appreciate your help. Thanks in advance -Sanjeev

1 0

Re: [Freeswitch-dev] (no subject)
by Brian West 23 Jan '08

23 Jan '08

FreeSWITCHers, I'm a little disappointed... 12 digs and one single comment? I know we can do better than that. Help spread the word! Thanks, Brian West On Jan 21, 2008, at 2:49 PM, Brian West wrote: > Only 7 diggs? Come on guys... I know there are more out there that > can digg it! ;) > > /b > > On Jan 21, 2008, at 2:49 PM, Michael Collins wrote: > >> FYI, I updated the latest release date/version on the Wikipedia page >> so that people don’t think FS has been idle since beta1/August 21, >> 2007! >> >> BTW, Anthony, nice work on the HD telephony article. >> -MC >> >> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org >> ] On Behalf Of Anthony Minessale >> Sent: Sunday, January 20, 2008 11:36 AM >> To: freeswitch-users(a)lists.freeswitch.org >> Subject: [Freeswitch-users] HD Telephony, Secure Audio >> >> Now that we have 16k and Secure RTP working abd a RC1, >> it's time to spread the word a little. >> >> I cross-posted my story at digg >> >> http://digg.com/tech_news/HD_Telephony_HIP_or_Hype >> >> Check it out and digg it! >> >> >> 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 >> >> >> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. >> Try it now. >> _______________________________________________ >> 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

2 1