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woomera status
by Josip Djuricic 15 Jan '08

15 Jan '08

Hi, can anyone tell me current woomera status, can it work actually, or not? I know there's a work on another h323 endpoint, but the question is can it be made to work acctually, and if yes can anyone share some info? Thanks, J.

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broken builds after svn update.
by Michael Jerris 13 Jan '08

13 Jan '08

I removed some autotools generated files from tree so if you svn update an existing working copy you will either have to re-bootstrap and configure, or you will need to autoreconf in the libs/libedit dir and re-configure in that directory. This will only be an issue on people who update a current checkout. New checkouts will be fine. Mike

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Re: [Freeswitch-users] [Freeswitch-dev] Call Forward @ Freeswitch
by Irshad Ahamed 10 Jan '08

10 Jan '08

Yeah, It's working by configuring the dial plan. But, imagine a case where PBX will be handling some 1000 extensions. Is it not tiresome to modify the dial plan for configuring the Call forward for numerous extensions?? BR, Irshad. On Jan 10, 2008 2:37 PM, Brian West <brian.west(a)mac.com> wrote: > You do that with simple logic in the dialplan. Similar to how the > default config saves and stores uuid's of last calls. > > /b > > On Jan 10, 2008, at 1:40 AM, Irshad Ahamed wrote: > > > Do we have support for Call Forward (Unconditional/Busy/No Answer) @ > > Freeswitch? If so, how should I enable it? > > > > BR, > > Irshad. > > _______________________________________________ > > 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 > > > _______________________________________________ > 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 >

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Re: [Freeswitch-users] [Freeswitch-dev] Call Forward @ Freeswitch
by Anthony Minessale 10 Jan '08

10 Jan '08

We are a switch not a PBX. Implementing those features in each endpoint sip,tdm etc is even more tiresome. There are ways to make an extension that uses the caller-id of the fxs line to key into a database and save call forwarding prefs etc. I suggest you investigate how FreeSWITCH works more and you will find several methods to do what you wish. 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: Thursday, January 10, 2008 12:35:04 PM Subject: Re: [Freeswitch-users] [Freeswitch-dev] Call Forward @ Freeswitch Yeah, It's working by configuring the dial plan. But, imagine a case where PBX will be handling some 1000 extensions. Is it not tiresome to modify the dial plan for configuring the Call forward for numerous extensions?? BR, Irshad. On Jan 10, 2008 2:37 PM, Brian West <brian.west(a)mac.com> wrote: You do that with simple logic in the dialplan. Similar to how the default config saves and stores uuid's of last calls. /b On Jan 10, 2008, at 1:40 AM, Irshad Ahamed wrote: > Do we have support for Call Forward (Unconditional/Busy/No Answer) @ > Freeswitch? If so, how should I enable it? > > BR, > Irshad. > _______________________________________________ > 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 _______________________________________________ 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-users] Not able to make an outgoing Call through FXO....
by Anthony Minessale 10 Jan '08

10 Jan '08

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. _______________________________________________ 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://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Call Forward @ Freeswitch
by Irshad Ahamed 10 Jan '08

10 Jan '08

Do we have support for Call Forward (Unconditional/Busy/No Answer) @ Freeswitch? If so, how should I enable it? BR, Irshad.

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Configuration
by freeswitch-users＠lists.freeswitch.org 09 Jan '08

09 Jan '08

Hello I still got problems with the configuration, since the new layout was added. Today I've updated to latest trunk (config files too) and I've configured the system. I've only done the following changes: 1. I've set domain=mydomain.com in vars.xml 2. I've renamed sip_profiles/default to mydomain.com and nat to nat_mydomain.com 3. I've enabled mod_xml_curl and configured it to fetch dial plans 4. I've commented out mod_dialplan_asterisk and mod_iax 5. I've added my extensions to directory\default (copied brian's config and made appropriate changes) My sip gateway do not use authentication (and I therefore just use " number(a)mygateway.com" in the dialplan for external calls). The phones register by using mydomain.com as proxy/registrar (not by using an ip). The UA's can register fine and external outbound calls works gr8. Current problems: * I only get audio one way when dialing from one registered UA to another, check attached log internal_call.log * Inbound external calls do not work, i get [407][Proxy Authentication Required], check log external_call.log //Jonas

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Dial plan for FXS/FXO
by sanjeev.mudholkar 09 Jan '08

09 Jan '08

Hi, I have installed a TDM11 card and the zaptel drivers. Enabled the mod_openzap in modules.conf.xml Trying to configure a dialplan to route incoming SIP calls to FXS. I couldn't find any documentation nor examples involving FXO/FXS in dialplan. Can some one suggest me an example to do the same? Thanks in advance -Sanjeev

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Re: [Freeswitch-users] [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....
by Anthony Minessale 09 Jan '08

09 Jan '08

Please update the code to the latest revision where it should be working right. In the future could you please direct issues to http://jira.freeswitch.org so we can properly track them. thanks =) 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: Tuesday, January 8, 2008 11:53:35 AM Subject: Re: [Freeswitch-users] [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... Your interpretation is right. This is what I have observed while testing here. To be more precise, when you hangup the FXS completely, the first call doesnt get disconnected. BR, Irshad. On Jan 8, 2008 7:38 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: Can you clarify more so I am sure I understand? You use fxs to call an extension that leads to a sip phone. You press FLASH on fxs to get a dialtone and make another call. You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end? The current intended behavior is when you FLASH fxs you can make another call and switch back and forth with subsequent FLASH. But if the fxs is ONHOOK all the calls should end. We still need some more code to do other behaviors from fxs like transfer 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: " freeswitch-dev(a)lists.freeswitch.org" <freeswitch-dev(a)lists.freeswitch.org> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Monday, January 7, 2008 11:36:09 PM Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... Set-up is as below: 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + 1 FXO registered with SIPX. 2. SNOM300 registered with SIPX SNOM300: Kernel Version:snom300 linux 3.25 Application-Version:snom300-SIP 6.5.2 Rootfs-Version:snom300 jffs2 v3.36 Firmware-URL: Production Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG 3. GRANDSTREAM registered with SIPX. GRANDSTREAM: Product Model: GXP2000 Software : Program-- 1.1.4.18 Version : Bootloader-- 1.1.4.6 Procedure followed: 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on hold @ Freeswitch FXS and make new call to GRANDSTREAM. 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ FREESWITCH FXS. 3. The second call with GRANDSTREAM is being disconnected but the first call with SNOM300 is still alive at SNOM300. NOTE: Please find the Freeswitch logs and Ethereal caps @ http://jira.freeswitch.org/browse/MODENDP-74 BR, Irshad. Looking for last minute shopping deals? Find them fast with Yahoo! Search. _______________________________________________ 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-users] [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....
by Irshad Ahamed 08 Jan '08

08 Jan '08

Your interpretation is right. This is what I have observed while testing here. To be more precise, when you hangup the FXS completely, the first call doesnt get disconnected. BR, Irshad. On Jan 8, 2008 7:38 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: > Can you clarify more so I am sure I understand? > > You use fxs to call an extension that leads to a sip phone. > You press FLASH on fxs to get a dialtone and make another call. > You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end? > > The current intended behavior is when you FLASH fxs you can make > another call and switch back and forth with subsequent FLASH. But if > the fxs is ONHOOK all the calls should end. > > We still need some more code to do other behaviors from fxs like > transfer 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: "freeswitch-dev(a)lists.freeswitch.org" < > freeswitch-dev(a)lists.freeswitch.org> > To: freeswitch-dev(a)lists.freeswitch.org; > freeswitch-users(a)lists.freeswitch.org > Sent: Monday, January 7, 2008 11:36:09 PM > Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... > > Set-up is as below: > 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + > 1 FXO registered with SIPX. > 2. SNOM300 registered with SIPX > > SNOM300: > Kernel Version:snom300 linux 3.25 > Application-Version:snom300-SIP 6.5.2 > Rootfs-Version:snom300 jffs2 v3.36 > Firmware-URL: > Production > Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB > V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG > 3. GRANDSTREAM registered with SIPX. > > GRANDSTREAM: > Product Model: GXP2000 > Software : Program-- 1.1.4.18 > Version : Bootloader-- 1.1.4.6 > > > Procedure followed: > 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on > hold @ Freeswitch FXS and make new call to GRANDSTREAM. > 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ > FREESWITCH FXS. > 3. The second call with GRANDSTREAM is being disconnected but the first > call with SNOM300 is still alive at SNOM300. > > NOTE: Please find the Freeswitch logs and Ethereal caps @ > http://jira.freeswitch.org/browse/MODENDP-74 > > BR, > Irshad. > > > ------------------------------ > 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-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 > >

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