Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
scott.ellis at novatex...
Guest





PostPosted: Wed Jan 14, 2009 5:09 am    Post subject: [Freeswitch-users] Failed bridge in dial plan still answers Reply with 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
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
anthony.minessale at g...
Guest





PostPosted: Wed Jan 14, 2009 8:42 am    Post subject: [Freeswitch-users] Failed bridge in dial plan still answers Reply with 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
Back to top
scott.ellis at novatex...
Guest





PostPosted: Wed Jan 14, 2009 6:32 pm    Post subject: [Freeswitch-users] Failed bridge in dial plan still answers Reply with quote

Thanks Anthony, got to say I am hugely impressed with the software - I am another Asterisk refugee Smile

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
Quote:


_______________________________________________
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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services