[Freeswitch-users] Default Dial Plan: action application bridge

Deya M deya787 at gmail.com
Wed Jun 30 21:49:26 PDT 2010


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:

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> 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
>>>
>>> _______________________________________________
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>>> http://www.freeswitch.org
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>>
>>
>> _______________________________________________
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
> ---------- 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"/>
>      </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/$1"/>
> >
> >
> >
> > On 28 June 2010 19:45, Tim St. Pierre <fs-list at communicatefreely.net
> > <mailto:fs-list at communicatefreely.net>> wrote:
> >
> >     Hello list,
> >
> >     I would like to bridge a call to multiple SIP endpoints, but add
> >     different headers to each.
> >
> >     I'm not entirely sure how to do this.  I have no problem exporting a
> >     SIP header that does what I
> >     want for one destination, but I'm not sure how to set it for two.
> >
> >     My application is that I want two IP phones to ring - one with the
> >     internal ring-ring splash, the
> >     others with a group-answer (single ring, then lamp flash only), for
> >     administrative assistants, etc.
> >
> >     How do I export different variables to each branch?
> >
> >     Thanks!
> >
> >     -Tim
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
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>
>
>
> ---------- Forwarded message ----------
> From: Deya M <deya787 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 1 Jul 2010 06:19:51 +0300
> Subject: [Freeswitch-users] Default Dial Plan: action application bridge
> Hi,
>
> In the default dial plan, with two extensions defined, 1000 and 1001, when
> I call from 1000 to 1001, I always get Voicemail, using the default config
> files: conf/dialplan/default.xml
>
> I changed the following from :
>
> * <action application="bridge" data="user/${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
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> http://www.freeswitch.org
>
>
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