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scott.ellis at novatex... Guest
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Posted: Wed Jan 14, 2009 5:09 am Post subject: [Freeswitch-users] Failed bridge in dial plan still answers |
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I have an inbound call via OpenZap, when I attempt to bridge to a SIP
extension, I get the ring tone (inbound line) up until the bridge fails
(for timeout or do not disturb). At this point the call is answered and
then my dial plan moves on to attempt another bridge to different
extensions. So I no longer have the ring tone for the person dialing in.
The call can still be answered and everything works ok, but I would
rather not answer the call until someone actually picks up. Failing that
simulating a ring tone would be good enough.
Have searched around, but at a bit of a loss on how to dothis.
Any suggestions greatly appreciated.
Scott
From my dialplan
<extension name="LandLine IN">
<condition field="source" expression="mod_openzap"/>
<condition field="caller_id_number" expression="^[1-8]$">
<!-- Ring reception for 30 seconds -->
<!--<action application="set" data="call_timeout=30"/> -->
<action application="set" data="continue_on_fail=true"/>
<!--<action application="set" data="hangup_after_bridge=true"/>-->
<action application="bridge"
data="{leg_timeout=30}sofia/$${domain}/500"/>
<!--<action application="playback"
data="sounds/ReceptionBusy.wav"/> -->
<!-- Ring second group for 15 seconds -->
<action application="set" data="call_timeout=15"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="ring_ready"/>
<action application="bridge"
data="${group_call(ringgroup2@${domain_name})"/>
<!-- Ring everybody -->
<action application="set" data="call_timeout=15"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="${group_call(everyone@${domain_name})"/>
<action application="hangup" data="NO_ANSWER"/>
</condition>
</extension>
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anthony.minessale at g... Guest
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Posted: Wed Jan 14, 2009 8:42 am Post subject: [Freeswitch-users] Failed bridge in dial plan still answers |
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Have a look here:
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
On Wed, Jan 14, 2009 at 4:03 AM, Scott Ellis <scott.ellis@novatex.com.au (scott.ellis@novatex.com.au)> wrote:
Quote: | I have an inbound call via OpenZap, when I attempt to bridge to a SIP
extension, I get the ring tone (inbound line) up until the bridge fails
(for timeout or do not disturb). At this point the call is answered and
then my dial plan moves on to attempt another bridge to different
extensions. So I no longer have the ring tone for the person dialing in.
The call can still be answered and everything works ok, but I would
rather not answer the call until someone actually picks up. Failing that
simulating a ring tone would be good enough.
Have searched around, but at a bit of a loss on how to dothis.
Any suggestions greatly appreciated.
Scott
From my dialplan
<extension name="LandLine IN">
<condition field="source" expression="mod_openzap"/>
<condition field="caller_id_number" expression="^[1-8]$">
<!-- Ring reception for 30 seconds -->
<!--<action application="set" data="call_timeout=30"/> -->
<action application="set" data="continue_on_fail=true"/>
<!--<action application="set" data="hangup_after_bridge=true"/>-->
<action application="bridge"
data="{leg_timeout=30}sofia/$${domain}/500"/>
<!--<action application="playback"
data="sounds/ReceptionBusy.wav"/> -->
<!-- Ring second group for 15 seconds -->
<action application="set" data="call_timeout=15"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="ring_ready"/>
<action application="bridge"
data="${group_call(ringgroup2@${domain_name})"/>
<!-- Ring everybody -->
<action application="set" data="call_timeout=15"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="${group_call(everyone@${domain_name})"/>
<action application="hangup" data="NO_ANSWER"/>
</condition>
</extension>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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scott.ellis at novatex... Guest
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Posted: Wed Jan 14, 2009 6:32 pm Post subject: [Freeswitch-users] Failed bridge in dial plan still answers |
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Thanks Anthony, got to say I am hugely impressed with the software - I am another Asterisk refugee
So the answering of the call even though the bridge fails is correct operation for the system? (Just curious)
Scott
Anthony Minessale wrote: Quote: | Have a look here:
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
On Wed, Jan 14, 2009 at 4:03 AM, Scott Ellis <scott.ellis@novatex.com.au (scott.ellis@novatex.com.au)> wrote:
Quote: | I have an inbound call via OpenZap, when I attempt to bridge to a SIP
extension, I get the ring tone (inbound line) up until the bridge fails
(for timeout or do not disturb). At this point the call is answered and
then my dial plan moves on to attempt another bridge to different
extensions. So I no longer have the ring tone for the person dialing in.
The call can still be answered and everything works ok, but I would
rather not answer the call until someone actually picks up. Failing that
simulating a ring tone would be good enough.
Have searched around, but at a bit of a loss on how to dothis.
Any suggestions greatly appreciated.
Scott
From my dialplan
<extension name="LandLine IN">
<condition field="source" expression="mod_openzap"/>
<condition field="caller_id_number" expression="^[1-8]$">
<!-- Ring reception for 30 seconds -->
<!--<action application="set" data="call_timeout=30"/> -->
<action application="set" data="continue_on_fail=true"/>
<!--<action application="set" data="hangup_after_bridge=true"/>-->
<action application="bridge"
data="{leg_timeout=30}sofia/$${domain}/500"/>
<!--<action application="playback"
data="sounds/ReceptionBusy.wav"/> -->
<!-- Ring second group for 15 seconds -->
<action application="set" data="call_timeout=15"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="ring_ready"/>
<action application="bridge"
data="${group_call(ringgroup2@${domain_name})"/>
<!-- Ring everybody -->
<action application="set" data="call_timeout=15"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="${group_call(everyone@${domain_name})"/>
<action application="hangup" data="NO_ANSWER"/>
</condition>
</extension>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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