[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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
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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
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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> IRC: irc.freenode.net #freeswitch
>
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> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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> 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"/>
> </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
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> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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