[Freeswitch-users] create a external FXO Trunk - A200
Deya M
deya787 at gmail.com
Wed Jun 30 23:16:36 PDT 2010
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 at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> 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 <deya787 at gmail.com>
> To: freeswitch-users at 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 at lists.freeswitch.org> wrote:
>
>> Send FreeSWITCH-users mailing list submissions to
>> freeswitch-users at lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>> freeswitch-users-request at lists.freeswitch.org
>>
>> You can reach the person managing the list at
>> freeswitch-users-owner at lists.freeswitch.org
>>
>> 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 at gmail.com>
>> To: freeswitch-users at 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 at 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 at celliax.org> wrote:
>>>
>>>> On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan <sameer2k3t at gmail.com>
>>>> wrote:
>>>> > Mr Giovani what was that ?
>>>>
>>>> just jokin, nevermind
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> ---------- Forwarded message ----------
>> From: Vladimir Elizarov <xengelpublicx at gmail.com>
>> To: freeswitch-users at 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 at communicatefreely.net>
>> To: freeswitch-users at 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 at communicatefreely.net
>> > <mailto:fs-list at 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
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Deya M <deya787 at gmail.com>
>> To: freeswitch-users at 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} ${dialed_extension}"/>
>> </condition>
>> </extension>
>>
>> :-D
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Wed, 30 Jun 2010 22:37:15 -0500
>> Subject: Re: [Freeswitch-users] Default Dial Plan: action application
>> bridge
>> you must have changed more than you think.
>>
>> you might want to diff you configs against the in-tree ones.
>>
>>
>> On Wed, Jun 30, 2010 at 10:19 PM, Deya M <deya787 at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> In the default dial plan, with two extensions defined, 1000 and 1001,
>>> when I call from 1000 to 1001, I always get Voicemail, using the default
>>> config files: conf/dialplan/default.xml
>>>
>>> I changed the following from :
>>>
>>> * <action application="bridge" data="user/${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 at 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
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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