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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