Re: [Freeswitch-users] create a external FXO Trunk - A200
Hi, Managed to ge I added the following to the dialplan: <extension name="OutgoingNumber"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bridge" data="openzap/1/4/${dialed_ext}"/> </condition> </extension> No ring. I get the following: 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:4/ 2010-07-01 11:07:04.627394 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:4/ [3ca74e5b-7e26-4a0a-b5f7-72d7a2f33fbf] 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:4/) State Change CS_NEW -> CS_INIT 2010-07-01 11:07:04.627394 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:59 Changing state on 1:4 from DOWN to DIALING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting. 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_INIT 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:431 (OpenZAP/1:4/) State Change CS_INIT -> CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_channel.c:1471 (OpenZAP/1:4/) Callstate Change DOWN -> RINGING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:4 2010-07-01 11:07:04.628392 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DIALING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:454 OpenZAP/1:4/ CHANNEL ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:4/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA going to sleep 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 2 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 1 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:4] STATE [DIALING] 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:838 Changing state on 1:4 from DIALING to DOWN 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DOWN 2010-07-01 11:07:21.166199 [DEBUG] mod_openzap.c:1556 got FXO sig 1:4 [STOP] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2261 (OpenZAP/1:4/) Callstate Change RINGING -> HANGUP 2010-07-01 11:07:21.166199 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:4/ [CS_CONSUME_MEDIA] [NONE] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:4/ [KILL] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_HANGUP 2010-07-01 11:07:21.166199 [DEBUG] zap_io.c:1388 channel done 1:4 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:4 thread ended. 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP 2010-07-01 11:07:21.168191 [WARNING] mod_openzap.c:517 VETO Changing state on 1:4 from DOWN to HANGUP 2010-07-01 11:07:21.168191 [DEBUG] mod_openzap.c:556 OpenZAP/1:4/ CHANNEL HANGUP 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:4/ Standard HANGUP, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-07-01 11:07:21.168191 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:4/) State Change CS_HANGUP -> CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:4/ Standard REPORTING, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:4/) State Change CS_REPORTING -> CS_DESTROY 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1175 Session 71 (OpenZAP/1:4/) Locked, Waiting on external entities 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1193 Session 71 (OpenZAP/1:4/) Ended 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:4/ [CS_DESTROY] But no ringing is heard. thanks, -:D On Thu, Jul 1, 2010 at 2:49 PM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: Default Dial Plan: action application bridge (Deya M)
---------- Forwarded message ---------- From: Deya M <deya787@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Thu, 1 Jul 2010 14:49:26 +1000 Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge I generated samples, and started from scratch.
I also made a copy of the conf after installation, rm it, and copied the original conf before testing!
-:D
On Thu, Jul 1, 2010 at 1:37 PM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:
1. Re: Query !! freeswitch, dingaling, jingle, skypopen (Anthony Minessale) 2. continue_on_fail and hangup_after_bridge with transfer (Vladimir Elizarov) 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) 4. Default Dial Plan: action application bridge (Deya M) 5. Re: Default Dial Plan: action application bridge (Anthony Minessale)
---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Wed, 30 Jun 2010 17:40:27 -0500 Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen If he didn't like to joke, he'd never survive in telephony.
On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan <sameer2k3t@gmail.com>wrote:
no problem. but did't expect from the creator of mod_skypopen, the wonder i would say
On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli < gmaruzz@celliax.org> wrote:
On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan <sameer2k3t@gmail.com> wrote:
Mr Giovani what was that ?
just jokin, nevermind
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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---------- Forwarded message ---------- From: Vladimir Elizarov <xengelpublicx@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Thu, 1 Jul 2010 03:29:09 +0400 Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with transfer I'm trying to make a dialplan:
<extension name="Long Dial"> <condition field="destination_number"
expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> <action application="set" data="real_dialed_number=7777777"/> </condition>
<condition field="destination_number" expression="^(79|89)(\d{9})$" break="on-true"> <action application="set" data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> <action application="bridge"
data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> <action application="transfer" data="to_card"/> </condition> </extension>
<extension name="to_card"> <condition field="destination_number" expression="^to_card$"> <action application="bridge" data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${ real_dialed_number}@192.168.50.53:5061<http://real_dialed_number%7D%40192.168.50.53%3A5061> "/> </condition> </extension>
The logic of his work: if unavailable gateway to the next. if not available all the gateway to go to the extension transfer. A problem in If we get a code busy here, it is not satisfied hangup_after_bridge. The call goes to the transfer. Why is this so?
trace: http://pastebin.freeswitch.org/13244
-- Best regards, Vladimir Elizarov
---------- Forwarded message ---------- From: "Tim St. Pierre" <fs-list@communicatefreely.net> To: freeswitch-users@lists.freeswitch.org Date: Wed, 30 Jun 2010 22:45:19 -0400 Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge Thanks all,
I read that bit, but didn't fully understand it's implications.
I'll give that a go.
-Tim
Steven Ayre wrote:
<action application="bridge"
data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/>
On 28 June 2010 19:45, Tim St. Pierre <fs-list@communicatefreely.net <mailto:fs-list@communicatefreely.net>> wrote:
Hello list,
I would like to bridge a call to multiple SIP endpoints, but add different headers to each.
I'm not entirely sure how to do this. I have no problem exporting a SIP header that does what I want for one destination, but I'm not sure how to set it for two.
My application is that I want two IP phones to ring - one with the internal ring-ring splash, the others with a group-answer (single ring, then lamp flash only), for administrative assistants, etc.
How do I export different variables to each branch?
Thanks!
-Tim
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---------- Forwarded message ---------- From: Deya M <deya787@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Thu, 1 Jul 2010 06:19:51 +0300 Subject: [Freeswitch-users] Default Dial Plan: action application bridge Hi,
In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml
I changed the following from :
* <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/>* TO *<action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/>*
and it did work! Not sure if the first / default one, * <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should work, but something is missing ?
New: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
Old:
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
:-D
---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Date: Wed, 30 Jun 2010 22:37:15 -0500 Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge you must have changed more than you think.
you might want to diff you configs against the in-tree ones.
On Wed, Jun 30, 2010 at 10:19 PM, Deya M <deya787@gmail.com> wrote:
Hi,
In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml
I changed the following from :
* <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/>* TO *<action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/>*
and it did work! Not sure if the first / default one, * <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should work, but something is missing ?
New: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
Old:
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension>
:-D
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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I may be wrong, but it looks like it is not detecting the dial tone, and therefore not dailing the number. You should check that connection to the line is correct, and that the tones are correctly configured for the carrier you use. François On Thu, 1 Jul 2010 16:16:36 +1000, Deya M wrote: Hi, Managed to ge I added the following to the dialplan: No ring. I get the following: 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:4/ 2010-07-01 11:07:04.627394 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:4/ [3ca74e5b-7e26-4a0a-b5f7-72d7a2f33fbf] 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:4/) State Change CS_NEW -> CS_INIT 2010-07-01 11:07:04.627394 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:59 Changing state on 1:4 from DOWN to DIALING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting. 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_INIT 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:431 (OpenZAP/1:4/) State Change CS_INIT -> CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_channel.c:1471 (OpenZAP/1:4/) Callstate Change DOWN -> RINGING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:4 2010-07-01 11:07:04.628392 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DIALING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:454 OpenZAP/1:4/ CHANNEL ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:4/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA going to sleep 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 2 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 1 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:4] STATE [DIALING] 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:838 Changing state on 1:4 from DIALING to DOWN 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DOWN 2010-07-01 11:07:21.166199 [DEBUG] mod_openzap.c:1556 got FXO sig 1:4 [STOP] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2261 (OpenZAP/1:4/) Callstate Change RINGING -> HANGUP 2010-07-01 11:07:21.166199 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:4/ [CS_CONSUME_MEDIA] [NONE] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:4/ [KILL] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_HANGUP 2010-07-01 11:07:21.166199 [DEBUG] zap_io.c:1388 channel done 1:4 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:4 thread ended. 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP 2010-07-01 11:07:21.168191 [WARNING] mod_openzap.c:517 VETO Changing state on 1:4 from DOWN to HANGUP 2010-07-01 11:07:21.168191 [DEBUG] mod_openzap.c:556 OpenZAP/1:4/ CHANNEL HANGUP 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:4/ Standard HANGUP, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-07-01 11:07:21.168191 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:4/) State Change CS_HANGUP -> CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:4/ Standard REPORTING, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:4/) State Change CS_REPORTING -> CS_DESTROY 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1175 Session 71 (OpenZAP/1:4/) Locked, Waiting on external entities 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1193 Session 71 (OpenZAP/1:4/) Ended 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:4/ [CS_DESTROY] But no ringing is heard. thanks, -:D On Thu, Jul 1, 2010 at 2:49 PM, wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org [2] To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org [4] You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org [5] When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Default Dial Plan: action application bridge (Deya M) ---------- Forwarded message ---------- From: Deya M To: freeswitch-users@lists.freeswitch.org [7] Date: Thu, 1 Jul 2010 14:49:26 +1000 Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge I generated samples, and started from scratch. I also made a copy of the conf after installation, rm it, and copied the original conf before testing! -:D On Thu, Jul 1, 2010 at 1:37 PM, wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org [9] To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [10] or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org [11] You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org [12] When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Query !! freeswitch, dingaling, jingle, skypopen (Anthony Minessale) 2. continue_on_fail and hangup_after_bridge with transfer (Vladimir Elizarov) 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) 4. Default Dial Plan: action application bridge (Deya M) 5. Re: Default Dial Plan: action application bridge (Anthony Minessale) ---------- Forwarded message ---------- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org [14] Date: Wed, 30 Jun 2010 17:40:27 -0500 Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, skypopen If he didn't like to joke, he'd never survive in telephony. On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan wrote: no problem. but did't expect from the creator of mod_skypopen, the wonder i would say On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli wrote: On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan wrote:
Mr Giovani what was that ?
just jokin, nevermind -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [22] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [23] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [24] http://www.freeswitch.org [25] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [26] ClueCon http://www.cluecon.com/ [27] Twitter: http://twitter.com/FreeSWITCH_wire [28] AIM: anthm MSN:anthony_minessale@hotmail.com [29] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [30] IRC: irc.freenode.net [31] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [32] googletalk:conf+888@conference.freeswitch.org [33] pstn:+19193869900 ---------- Forwarded message ---------- From: Vladimir Elizarov To: freeswitch-users@lists.freeswitch.org [35] Date: Thu, 1 Jul 2010 03:29:09 +0400 Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with transfer I'm trying to make a dialplan: expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^+([0-9]{11})$)$"> data="hangup_after_bridge=true"/> application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> break="on-true"> data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> data="{execute_on_answer=send_dtmfs${dtmf}}sofia/internal/${real_dialed_number}@192.168.50.53:5061 [36]"/> The logic of his work: if unavailable gateway to the next. if not available all the gateway to go to the extension transfer. A problem in If we get a code busy here, it is not satisfied hangup_after_bridge. The call goes to the transfer. Why is this so? trace: http://pastebin.freeswitch.org/13244 [37] -- Best regards, Vladimir Elizarov ---------- Forwarded message ---------- From: "Tim St. Pierre" To: freeswitch-users@lists.freeswitch.org [39] Date: Wed, 30 Jun 2010 22:45:19 -0400 Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge Thanks all, I read that bit, but didn't fully understand it's implications. I'll give that a go. -Tim Steven Ayre wrote:
data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/>
On 28 June 2010 19:45, Tim St. Pierre
fs-list@communicatefreely.net [41]>> wrote:
Hello list,
I would
like to bridge a call to multiple SIP endpoints, but add
different headers to each.
I'm not entirely sure how to do this. I have no problem exporting a SIP header that does what I want for one destination, but I'm not sure how to set it for two.
My application is that I want two IP phones to ring - one with the internal ring-ring splash, the others with a group-answer (single ring, then lamp flash only), for administrative assistants, etc.
How do I export different variables to each branch?
Thanks!
-Tim
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---------- Forwarded message ---------- From: Deya M To: freeswitch-users@lists.freeswitch.org [52] Date: Thu, 1 Jul 2010 06:19:51 +0300 Subject: [Freeswitch-users] Default Dial Plan: action application bridge Hi, In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml I changed the following from : TO and it did work! Not sure if the first / default one, should work, but something is missing ? New: Old: :-D ---------- Forwarded message ---------- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org [54] Date: Wed, 30 Jun 2010 22:37:15 -0500 Subject: Re: [Freeswitch-users] Default Dial Plan: action application bridge you must have changed more than you think. you might want to diff you configs against the in-tree ones. On Wed, Jun 30, 2010 at 10:19 PM, Deya M wrote: Hi, In the default dial plan, with two extensions defined, 1000 and 1001, when I call from 1000 to 1001, I always get Voicemail, using the default config files: conf/dialplan/default.xml I changed the following from : TO and it did work! Not sure if the first / default one, should work, but something is missing ? New: Old: :-D _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [56] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [57] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [58] http://www.freeswitch.org [59] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [60] ClueCon http://www.cluecon.com/ [61] Twitter: http://twitter.com/FreeSWITCH_wire [62] AIM: anthm MSN:anthony_minessale@hotmail.com [63] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [64] IRC: irc.freenode.net [65] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [66] googletalk:conf+888@conference.freeswitch.org [67] pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [68] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [70] http://www.freeswitch.org [71] _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [72] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [73] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [74] http://www.freeswitch.org [75] Links: ------ [1] mailto:freeswitch-users-request@lists.freeswitch.org [2] mailto:freeswitch-users@lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] mailto:freeswitch-users-request@lists.freeswitch.org [5] mailto:freeswitch-users-owner@lists.freeswitch.org [6] mailto:deya787@gmail.com [7] mailto:freeswitch-users@lists.freeswitch.org [8] mailto:freeswitch-users-request@lists.freeswitch.org [9] mailto:freeswitch-users@lists.freeswitch.org [10] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [11] mailto:freeswitch-users-request@lists.freeswitch.org [12] mailto:freeswitch-users-owner@lists.freeswitch.org [13] mailto:anthony.minessale@gmail.com [14] mailto:freeswitch-users@lists.freeswitch.org [15] mailto:sameer2k3t@gmail.com [16] mailto:gmaruzz@celliax.org [17] mailto:sameer2k3t@gmail.com [18] mailto:FreeSWITCH-users@lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] mailto:FreeSWITCH-users@lists.freeswitch.org [23] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [24] http://lists.freeswitch.org/mailman/options/freeswitch-users [25] http://www.freeswitch.org [26] http://www.freeswitch.org/ [27] http://www.cluecon.com/ [28] http://twitter.com/FreeSWITCH_wire [29] mailto:MSN%3Aanthony_minessale@hotmail.com [30] mailto:PAYPAL%3Aanthony.minessale@gmail.com [31] http://irc.freenode.net [32] mailto:sip%3A888@conference.freeswitch.org [33] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org [34] mailto:xengelpublicx@gmail.com [35] mailto:freeswitch-users@lists.freeswitch.org [36] http://real_dialed_number%7D%40192.168.50.53%3A5061 [37] http://pastebin.freeswitch.org/13244 [38] mailto:fs-list@communicatefreely.net [39] mailto:freeswitch-users@lists.freeswitch.org [40] mailto:fs-list@communicatefreely.net [41] mailto:fs-list@communicatefreely.net [42] mailto:FreeSWITCH-users@lists.freeswitch.org [43] mailto:FreeSWITCH-users@lists.freeswitch.org [44] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [45] http://lists.freeswitch.org/mailman/options/freeswitch-users [46] http://www.freeswitch.org [47] mailto:FreeSWITCH-users@lists.freeswitch.org [48] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [49] http://lists.freeswitch.org/mailman/options/freeswitch-users [50] http://www.freeswitch.org [51] mailto:deya787@gmail.com [52] mailto:freeswitch-users@lists.freeswitch.org [53] mailto:anthony.minessale@gmail.com [54] mailto:freeswitch-users@lists.freeswitch.org [55] mailto:deya787@gmail.com [56] mailto:FreeSWITCH-users@lists.freeswitch.org [57] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [58] http://lists.freeswitch.org/mailman/options/freeswitch-users [59] http://www.freeswitch.org [60] http://www.freeswitch.org/ [61] http://www.cluecon.com/ [62] http://twitter.com/FreeSWITCH_wire [63] mailto:MSN%3Aanthony_minessale@hotmail.com [64] mailto:PAYPAL%3Aanthony.minessale@gmail.com [65] http://irc.freenode.net [66] mailto:sip%3A888@conference.freeswitch.org [67] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org [68] mailto:FreeSWITCH-users@lists.freeswitch.org [69] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [70] http://lists.freeswitch.org/mailman/options/freeswitch-users [71] http://www.freeswitch.org [72] mailto:FreeSWITCH-users@lists.freeswitch.org [73] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [74] http://lists.freeswitch.org/mailman/options/freeswitch-users [75] http://www.freeswitch.org
participants (2)
-
Deya M
-
François Legal