Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have: <context name="vpbx"> <extension name="local_number"> ... <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::$${base_dir}/recordings/${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="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension> So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf: features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ... To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great. 1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature. I'm using FreeSWITCH Version 1.0.trunk (14633M) Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here. Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( . Thank you in advance, Anatoliy Kounitskiy
After several hours of testing I was able to answer myself the previous mentioned questions. It appears that # and the 0 option work _only_ if user C has answered the call OR voicemail system answers it. user A ---call---> user B----attended xfer---> user C At this point I have new question. In example user C does not have a voicemail and the call timeout is not an option to wait for. How can user B go back to the user A, who is listening to MOH? Could someone help me with an advice/tip? At the moment I have just one idea for accomplishing it: 1) try to use bind_meta_app in the extension with the att_xfer (not sure if it can be done). To have a key feature that takes the user A call leg id and bridging it with user B Thank you in advnace, Anatoliy Kounitskiy On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy Kounitskiy<anatoliy@kounitskiy.com> wrote:
Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have:
<context name="vpbx"> <extension name="local_number"> ... <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::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension>
So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf:
features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ...
To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great.
1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature.
I'm using FreeSWITCH Version 1.0.trunk (14633M)
Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here.
Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( .
Thank you in advance, Anatoliy Kounitskiy
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
maybe we can make an origination_cancel_key=# you could set on the dial string to be able to cancel that originate with dtmf On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy < anatoliy@kounitskiy.com> wrote:
After several hours of testing I was able to answer myself the previous mentioned questions.
It appears that # and the 0 option work _only_ if user C has answered the call OR voicemail system answers it.
user A ---call---> user B----attended xfer---> user C
At this point I have new question. In example user C does not have a voicemail and the call timeout is not an option to wait for. How can user B go back to the user A, who is listening to MOH? Could someone help me with an advice/tip?
At the moment I have just one idea for accomplishing it: 1) try to use bind_meta_app in the extension with the att_xfer (not sure if it can be done). To have a key feature that takes the user A call leg id and bridging it with user B
Thank you in advnace, Anatoliy Kounitskiy
On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy Kounitskiy<anatoliy@kounitskiy.com> wrote:
Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have:
<context name="vpbx"> <extension name="local_number"> ... <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::$${base_dir}/recordings/${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="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension>
So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf:
features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ...
To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great.
1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature.
I'm using FreeSWITCH Version 1.0.trunk (14633M)
Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here.
Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( .
Thank you in advance, Anatoliy Kounitskiy
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
-- 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
i added a patch to attempt to do this so try adding {origination_cancel_key=#} before the dial string or <action application="set" data="origination_cancel_key=#"/> before you bridge. On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
maybe we can make an origination_cancel_key=# you could set on the dial string to be able to cancel that originate with dtmf
On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy < anatoliy@kounitskiy.com> wrote:
After several hours of testing I was able to answer myself the previous mentioned questions.
It appears that # and the 0 option work _only_ if user C has answered the call OR voicemail system answers it.
user A ---call---> user B----attended xfer---> user C
At this point I have new question. In example user C does not have a voicemail and the call timeout is not an option to wait for. How can user B go back to the user A, who is listening to MOH? Could someone help me with an advice/tip?
At the moment I have just one idea for accomplishing it: 1) try to use bind_meta_app in the extension with the att_xfer (not sure if it can be done). To have a key feature that takes the user A call leg id and bridging it with user B
Thank you in advnace, Anatoliy Kounitskiy
On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy Kounitskiy<anatoliy@kounitskiy.com> wrote:
Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have:
<context name="vpbx"> <extension name="local_number"> ... <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::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension>
So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf:
features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ...
To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great.
1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature.
I'm using FreeSWITCH Version 1.0.trunk (14633M)
Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here.
Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( .
Thank you in advance, Anatoliy Kounitskiy
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
-- 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
Thank you for the fast patch :) I'm going to test it right away and report you if there are problems or not. On Wed, Aug 26, 2009 at 10:59 PM, Anthony Minessale<anthony.minessale@gmail.com> wrote:
i added a patch to attempt to do this so try adding
{origination_cancel_key=#} before the dial string or <action application="set" data="origination_cancel_key=#"/> before you bridge.
On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
maybe we can make an origination_cancel_key=# you could set on the dial string to be able to cancel that originate with dtmf
On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
After several hours of testing I was able to answer myself the previous mentioned questions.
It appears that # and the 0 option work _only_ if user C has answered the call OR voicemail system answers it.
user A ---call---> user B----attended xfer---> user C
At this point I have new question. In example user C does not have a voicemail and the call timeout is not an option to wait for. How can user B go back to the user A, who is listening to MOH? Could someone help me with an advice/tip?
At the moment I have just one idea for accomplishing it: 1) try to use bind_meta_app in the extension with the att_xfer (not sure if it can be done). To have a key feature that takes the user A call leg id and bridging it with user B
Thank you in advnace, Anatoliy Kounitskiy
On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy Kounitskiy<anatoliy@kounitskiy.com> wrote:
Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have:
<context name="vpbx"> <extension name="local_number"> ... <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::$${base_dir}/recordings/${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="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension>
So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf:
features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ...
To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great.
1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature.
I'm using FreeSWITCH Version 1.0.trunk (14633M)
Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here.
Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( .
Thank you in advance, Anatoliy Kounitskiy
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
-- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ 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
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
i added a patch to attempt to do this so try adding
Nice work!
{origination_cancel_key=#} before the dial string or <action application="set" data="origination_cancel_key=#"/> before you bridge.
Anatoliy, Please try this and let us know if it works for you as expected. If so, please write up the exact procedure that you used and include dialplan examples. I'd like to get this on the wiki while it's fresh in my mind. If you feel comfortable editing the wiki on your own then just let me know so that we can coordinate efforts. Thanks, MC
Just for information - the idea : A---calls---> B ---att_xfer---> C , B hangs C up and goes back to A ( I`m sorry C is not answering :) ) dialplan: ----------- <extension name="local_number"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <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::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=USER_NOT_REGISTERED,SUBSCRIBER_ABSENT"/> <action application="info"/> <action application="set" data="bringback=${user_data(${dialed_extension}@${domain_name} var cringback )}"/> <action application="set" data="ringback=${${bringback}}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="info"/> ... </condition> </extension> features.xml --------------------- <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_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="bringback=${user_data(${attxfer_callthis}@${domain_name} var cringback )}"/> <action application="set" data="ringback=${${bringback}}"/> <action application="info"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> After testing the patch the results are: 1) B answers the call from A and executes the feature code *4 (attented_xfer), sends the desired number ( in this case the number of C). A goes to MusicOnHold, B is waiting for C to pick up - B sends # - sends SIP CANCEL to C and SIP BUY to A - so all calls are dropped. 2) B answers the call from A and executes the feature code *4 (attented_xfer), but without entering the number of C - so the read aplication timeouts after 30 sec - after that att_xfer is executed with empty string (nothing is entered) - B and A are bridged together. On the console without entering the extension of C: 2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute set(origination_cancel_key=#) EXECUTE sofia/internal/sip:102@10.17.4.107:5060 set(origination_cancel_key=#) 2009-08-27 16:00:58.929756 [DEBUG] mod_dptools.c:748 sofia/internal/sip:userB@10.17.4.107:5060 SET [origination_cancel_key]=[#] 2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute att_xfer(user/${attxfer_callthis}@${domain_name}) EXECUTE sofia/internal/sip:102@10.17.4.107:5060 att_xfer(user/@1.1.1.1) 2009-08-27 16:00:58.929756 [WARNING] mod_dptools.c:2373 Can't find user [@1.1.1.1] 2009-08-27 16:00:58.929756 [ERR] switch_ivr_originate.c:1527 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-08-27 16:00:58.929756 [DEBUG] switch_ivr_originate.c:2167 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_play_say.c:1402 done playing file 2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231 sofia/internal/userA@1.1.1.1 receive message [BRIDGE] 2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/userA@1.1.1.1 [BREAK] 2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:233 Send signal sofia/internal/sip:userB@10.17.4.107:5060 [BREAK] 2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231 sofia/internal/sip:userB@10.17.4.107:5060 receive message [BRIDGE] 2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/sip:userB@10.17.4.107:5060 [BREAK] 2009-08-27 16:00:58.954052 [DEBUG] switch_ivr_bridge.c:233 Send signal sofia/internal/userA@1.1.1.1 [BREAK] 2009-08-27 16:01:22.881501 [DEBUG] sofia.c:3302 Channel sofia/internal/userA@1.1.1.1 entering state [calling][0] 2009-08-27 16:01:22.886122 [DEBUG] sofia.c:3302 Channel sofia/internal/userA@1.1.1.1 entering state [ready][200] I'll try to summarize my idea - could it be possible when the # (in the att_xfer) is executed to behave as if you're trying to make attended transfer to non existing subscriber( SUBSCRIBER_ABSENT ), so B can be bridged back with A. Thank you in advance, Anatoliy Kounitskiy On Wed, Aug 26, 2009 at 11:11 PM, Michael Collins<msc@freeswitch.org> wrote:
On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
i added a patch to attempt to do this so try adding
Nice work!
{origination_cancel_key=#} before the dial string or <action application="set" data="origination_cancel_key=#"/> before you bridge.
Anatoliy,
Please try this and let us know if it works for you as expected. If so, please write up the exact procedure that you used and include dialplan examples. I'd like to get this on the wiki while it's fresh in my mind. If you feel comfortable editing the wiki on your own then just let me know so that we can coordinate efforts.
Thanks, MC
_______________________________________________ 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
please retest with latest SVN trunk On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy < anatoliy@kounitskiy.com> wrote:
Just for information - the idea : A---calls---> B ---att_xfer---> C , B hangs C up and goes back to A ( I`m sorry C is not answering :) )
-- 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
Thank you all for the great job :) Now it works as I wanted!! Tomorrow I'll will try to update the wiki :) But with few words ( I tried to strip the extensions to minimum for more easy read for users). In the dialplan I have created the extension showed below: <extension name="local_number"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::attented_xfer XML features"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> In the features.xml file I have extension attented_xfer, that is using the att_xfer application : <extension name="attented_xfer"> <condition field="destination_number" expression="^attented_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> Also I have set the the origination cancel key will be # - so if B wants to cancel the call to C and to go back to A - just uses the # key :) Again - thank you very much :) :) Anatoliy Kounitskiy On Thu, Aug 27, 2009 at 7:48 PM, Anthony Minessale<anthony.minessale@gmail.com> wrote:
please retest with latest SVN trunk
On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
Just for information - the idea : A---calls---> B ---att_xfer---> C , B hangs C up and goes back to A ( I`m sorry C is not answering :) )
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
Thanks! I'll be in touch to help you if you need any wiki editing assistance. -MC On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy < anatoliy@kounitskiy.com> wrote:
Thank you all for the great job :) Now it works as I wanted!!
Tomorrow I'll will try to update the wiki :)
But with few words ( I tried to strip the extensions to minimum for more easy read for users). In the dialplan I have created the extension showed below:
<extension name="local_number"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::attented_xfer XML features"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension>
In the features.xml file I have extension attented_xfer, that is using the att_xfer application :
<extension name="attented_xfer"> <condition field="destination_number" expression="^attented_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="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension>
Also I have set the the origination cancel key will be # - so if B wants to cancel the call to C and to go back to A - just uses the # key :)
Again - thank you very much :) :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 7:48 PM, Anthony Minessale<anthony.minessale@gmail.com> wrote:
please retest with latest SVN trunk
On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
Just for information - the idea : A---calls---> B ---att_xfer---> C , B hangs C up and goes back to A ( I`m sorry C is not answering :) )
-- 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
_______________________________________________ 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
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
The wiki page is updated ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope it is understandable :) Anatoliy Kounitskiy On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc@freeswitch.org> wrote:
Thanks! I'll be in touch to help you if you need any wiki editing assistance. -MC
On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
Thank you all for the great job :) Now it works as I wanted!!
Tomorrow I'll will try to update the wiki :)
But with few words ( I tried to strip the extensions to minimum for more easy read for users). In the dialplan I have created the extension showed below:
<extension name="local_number"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::attented_xfer XML features"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension>
In the features.xml file I have extension attented_xfer, that is using the att_xfer application :
<extension name="attented_xfer"> <condition field="destination_number" expression="^attented_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension>
Also I have set the the origination cancel key will be # - so if B wants to cancel the call to C and to go back to A - just uses the # key :)
Again - thank you very much :) :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 7:48 PM, Anthony Minessale<anthony.minessale@gmail.com> wrote:
please retest with latest SVN trunk
On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
Just for information - the idea : A---calls---> B ---att_xfer---> C , B hangs C up and goes back to A ( I`m sorry C is not answering :) )
-- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ 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
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
_______________________________________________ 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
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
Thanks! I made very minor changes. We appreciate you helping out with documenting new features. -MC On Fri, Aug 28, 2009 at 4:13 AM, Anatoliy Kounitskiy < anatoliy@kounitskiy.com> wrote:
The wiki page is updated ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope it is understandable :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc@freeswitch.org> wrote:
Thanks! I'll be in touch to help you if you need any wiki editing assistance. -MC
On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
Thank you all for the great job :) Now it works as I wanted!!
Tomorrow I'll will try to update the wiki :)
But with few words ( I tried to strip the extensions to minimum for more easy read for users). In the dialplan I have created the extension showed below:
<extension name="local_number"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::attented_xfer XML features"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension>
In the features.xml file I have extension attented_xfer, that is using the att_xfer application :
<extension name="attented_xfer"> <condition field="destination_number" expression="^attented_xfer$"> <action application="set" data="continue_on_fail=true"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension>
Also I have set the the origination cancel key will be # - so if B wants to cancel the call to C and to go back to A - just uses the # key :)
Again - thank you very much :) :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 7:48 PM, Anthony Minessale<anthony.minessale@gmail.com> wrote:
please retest with latest SVN trunk
On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com> wrote:
Just for information - the idea : A---calls---> B ---att_xfer---> C , B hangs C up and goes back to
A
( I`m sorry C is not answering :) )
-- 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
_______________________________________________ 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
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
_______________________________________________ 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
-- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy@kounitskiy.com Mobile: +359898913540
_______________________________________________ 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
participants (4)
-
Anatoliy Kounitskiy
-
Anthony Minessale
-
Brian West
-
Michael Collins