Hello, I have some problems with attended transfer and loopback Scenario how id <does> work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped How it <should> work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH) The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario. Here are the configs: In dialplan for calling B: <anti-action application="bind_meta_app" data="4 b b execute_extension::attended_xfer XML default"/> Dialplan for executing the att_xfer: <extension name="attended_xfer"> <condition field="destination_number" expression="^attended_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="loopback/${attxfer_callthis}"/> </condition> </extension> So this is pretty standard, except the loopback. SVN is 15322. Anybody has a solution for this? Best regards Peter
set/export the channel variable loopback_bowout=true so it's on the loopback leg On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id <does> work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped
How it <should> work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH)
The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario.
Here are the configs: In dialplan for calling B: <anti-action application="bind_meta_app" data="4 b b execute_extension::attended_xfer XML default"/>
Dialplan for executing the att_xfer: <extension name="attended_xfer"> <condition field="destination_number" expression="^attended_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="loopback/${attxfer_callthis}"/> </condition> </extension>
So this is pretty standard, except the loopback. SVN is 15322.
Anybody has a solution for this?
Best regards Peter
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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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
hit send too soon you want to set loopback_bowout=false This keeps loopback from trying to destroy itself when it sees a chance to cut out of the call path. On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
set/export the channel variable loopback_bowout=true so it's on the loopback leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX <Prometheus001@gmx.net>wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id <does> work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped
How it <should> work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH)
The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario.
Here are the configs: In dialplan for calling B: <anti-action application="bind_meta_app" data="4 b b execute_extension::attended_xfer XML default"/>
Dialplan for executing the att_xfer: <extension name="attended_xfer"> <condition field="destination_number" expression="^attended_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="loopback/${attxfer_callthis}"/> </condition> </extension>
So this is pretty standard, except the loopback. SVN is 15322.
Anybody has a solution for this?
Best regards Peter
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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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks Anthony, however this rather deteriorated the situation. Now it works the following - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C both have no voice (and B is dropped) - When A hangs up, C hangs up Before it did: - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped - When A hangs up, C hangs up Best regards Peter Anthony Minessale schrieb:
hit send too soon you want to set loopback_bowout=false
This keeps loopback from trying to destroy itself when it sees a chance to cut out of the call path.
On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
set/export the channel variable loopback_bowout=true so it's on the loopback leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id <does> work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped
How it <should> work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH)
The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario.
Here are the configs: In dialplan for calling B: <anti-action application="bind_meta_app" data="4 b b execute_extension::attended_xfer XML default"/>
Dialplan for executing the att_xfer: <extension name="attended_xfer"> <condition field="destination_number" expression="^attended_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="loopback/${attxfer_callthis}"/> </condition> </extension>
So this is pretty standard, except the loopback. SVN is 15322.
Anybody has a solution for this?
Best regards Peter
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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if you provide a console trace of both situations with console loglevel debug and put them on pastebin i can tell you what's happening. On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
Thanks Anthony,
however this rather deteriorated the situation. Now it works the following - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C both have no voice (and B is dropped) - When A hangs up, C hangs up
Before it did: - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped - When A hangs up, C hangs up
Best regards Peter
Anthony Minessale schrieb:
hit send too soon you want to set loopback_bowout=false
This keeps loopback from trying to destroy itself when it sees a chance to cut out of the call path.
On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
set/export the channel variable loopback_bowout=true so it's on the loopback leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id <does> work - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped
How it <should> work until here: - A calls B - B enters *4 gets an announcement and enter digits for C (A get MOH) - C is called - As soon as C picks up the call, B and C are connected (A still MOH)
The dial string for C is dynamic and dependent on certain parameters, therefore C must be called via Loopback in our scenario.
Here are the configs: In dialplan for calling B: <anti-action application="bind_meta_app" data="4 b b execute_extension::attended_xfer XML default"/>
Dialplan for executing the att_xfer: <extension name="attended_xfer"> <condition field="destination_number" expression="^attended_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="loopback/${attxfer_callthis}"/> </condition> </extension>
So this is pretty standard, except the loopback. SVN is 15322.
Anybody has a solution for this?
Best regards Peter
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
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IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
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pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
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Hello Anthony, I made a console trace today: http://pastebin.freeswitch.org/11125 Different from the mail below, in this case A and C have voice. Best regards Peter Anthony Minessale schrieb:
if you provide a console trace of both situations with console loglevel debug and put them on pastebin i can tell you what's happening.
On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
Thanks Anthony,
however this rather deteriorated the situation. Now it works the following - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C both have no voice (and B is dropped) - When A hangs up, C hangs up
Before it did: - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C are connected and B is dropped - When A hangs up, C hangs up
Best regards Peter
Anthony Minessale schrieb: > hit send too soon > you want to set loopback_bowout=false > > This keeps loopback from trying to destroy itself when it sees a > chance to cut out of the call path. > > > On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale > <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com> <mailto:anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>>> wrote: > > > set/export the channel variable loopback_bowout=true so it's on > the loopback leg > > > > > On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX > <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net> <mailto:Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>>> wrote: > > Hello, > > I have some problems with attended transfer and loopback > > Scenario how id <does> work > - A calls B > - B enters *4 gets an announcement and enter digits for C (A > get MOH) > - C is called > - As soon as C picks up the call, A and C are connected and B > is dropped > > How it <should> work until here: > - A calls B > - B enters *4 gets an announcement and enter digits for C (A > get MOH) > - C is called > - As soon as C picks up the call, B and C are connected (A > still MOH) > > The dial string for C is dynamic and dependent on certain > parameters, > therefore C must be called via Loopback in our scenario. > > > Here are the configs: > In dialplan for calling B: > <anti-action application="bind_meta_app" data="4 b b > execute_extension::attended_xfer XML default"/> > > Dialplan for executing the att_xfer: > <extension name="attended_xfer"> > <condition field="destination_number" > expression="^attended_xfer$"> > <action application="set" data="continue_on_fail=true"/> > <action application="read" data="3 4 ivr/ivr-enter_ext.wav > attxfer_callthis 30000 #"/> > <action application="set" data="origination_cancel_key=#"/> > <action application="att_xfer" > data="loopback/${attxfer_callthis}"/> > </condition> > </extension> > > So this is pretty standard, except the loopback. SVN is 15322. > > Anybody has a solution for this? > > > Best regards > Peter > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto: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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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participants (2)
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Anthony Minessale
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Peter P GMX