Hi, *Just i want to call the 1002==>9841799874 and i want to call another no 9840544078 and add in to the conference room * *First method i tried :* api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874@172.20.191.228) ====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Both caller and extension will be transfer into conference room Then i call the another caller api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9840544078@172.20.191.228)====> 1002 call another caller 9840544078 api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001 ====>Transfer only the caller 9840544078 to conference room 3001 while calling the second no 9840544078 the first line get hold after the transfer of bleg they all come in to the conference room but the first line caller is in hold so we cant able to unhold the call through api we have to manually go the soft phone and unhold the call . but i need the unhold should be done in api what is the solution . *i have tried in another method also * api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874@172.20.191.228)====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Transfer only the caller 9841799874 to conference room 3001 api uuid_kill 04120c08-19e7-4881-8873-0834a11524ff before i hangup i have transfer both the caller in to conference room so when i hangup the extension why my extension got hangup Can any one suggest me how can i try the conference are else correct where i am wrong. -- Warm Regards, N.Baskar
Does the second call to 9840544078 have to come from 1002? You can always just call 9840544078 and drop it into the conference directly: api originate sofia/default/9840544078@172.20.191.228 &conference(3001) or set the channel variable to simulate it coming from 1002 api originate {origination_caller_id_number=1002}sofia/default/ 9840544078@172.20.191.228 &conference(3001) Granted, not sure I completely understand your problem. --- danule On Wed, Nov 12, 2008 at 2:30 AM, Baskar <yudha2008@gmail.com> wrote:
Hi,
*Just i want to call the 1002==>9841799874 and i want to call another no 9840544078 and add in to the conference room *
*First method i tried :*
api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874@172.20.191.228) ====>1002 call 9841799874
api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Both caller and extension will be transfer into conference room
Then i call the another caller
api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9840544078@172.20.191.228)====> 1002 call another caller 9840544078
api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001 ====>Transfer only the caller 9840544078 to conference room 3001
while calling the second no 9840544078 the first line get hold
after the transfer of bleg they all come in to the conference room but the first line caller is in hold so we cant able to unhold the call through api we have to manually go the soft phone and unhold the call .
but i need the unhold should be done in api what is the solution .
*i have tried in another method also *
api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874@172.20.191.228)====>1002 call 9841799874
api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Transfer only the caller 9841799874 to conference room 3001
api uuid_kill 04120c08-19e7-4881-8873-0834a11524ff
before i hangup i have transfer both the caller in to conference room so when i hangup the extension why my extension got hangup
Can any one suggest me how can i try the conference are else correct where i am wrong.
-- Warm Regards, N.Baskar
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Hi, Conference is working well in recent days, but suddenly it in not working when i transfer the call ,the call get disconnected with the voice file playing Bye....... What is the error? When i transfer the call why the call get hangup api originate sofia/internal/1003%172.20.191.227 &bridge(sofia/default/ 79841799874@172.20.191.228) Content-Type: api/response Content-Length: 41 +OK b98ddfca-6e84-4945-a02c-e7d455b58c37 api uuid_transfer b98ddfca-6e84-4945-a02c-e7d455b58c37 -both 3003 Content-Type: api/response Content-Length: 4 +OK Correct me were i am Wrong. -- Warm Regards, N.Baskar
On Jan 2, 2009, at 9:58 AM, Baskar wrote:
Hi,
Conference is working well in recent days, but suddenly it in not working when i transfer the call ,the call get disconnected with the voice file playing Bye.......
What is the error? When i transfer the call why the call get hangup
api originate sofia/internal/1003%172.20.191.227 &bridge(sofia/default/79841799874@172.20.191.228 )
Content-Type: api/response Content-Length: 41
+OK b98ddfca-6e84-4945-a02c-e7d455b58c37 api uuid_transfer b98ddfca-6e84-4945-a02c-e7d455b58c37 -both 3003
Content-Type: api/response Content-Length: 4
+OK
Correct me were i am Wrong.
Look at the debug log to see what happens. Mike
I'm guessing you are doing it too fast before the call is completed. On Fri, Jan 2, 2009 at 11:56 AM, Michael Jerris <mike@jerris.com> wrote:
On Jan 2, 2009, at 9:58 AM, Baskar wrote:
Hi,
Conference is working well in recent days, but suddenly it in not working when i transfer the call ,the call get disconnected with the voice file playing Bye.......
What is the error? When i transfer the call why the call get hangup
api originate sofia/internal/1003%172.20.191.227 &bridge(sofia/default/ 79841799874@172.20.191.228)
Content-Type: api/response Content-Length: 41
+OK b98ddfca-6e84-4945-a02c-e7d455b58c37 api uuid_transfer b98ddfca-6e84-4945-a02c-e7d455b58c37 -both 3003
Content-Type: api/response Content-Length: 4
+OK
Correct me were i am Wrong.
Look at the debug log to see what happens.
Mike
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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
*Hi, Conference is working well before. Suddenly it is not working i have tested manually by calling 3002 conference room from Soft phone .But call get hangup directly with the ivr message BYE. I have given "^(30\d{2})$" in default.xml . what is the error? I have pasted the console logs in this path http://pastebin.freeswitch.org/6617 Try to correct where is error and what is the error???? -- Warm Regards, N.Baskar *
you might want to press f8 first so you actually get debug log. On Sat, Jan 3, 2009 at 3:44 AM, Baskar <yudha2008@gmail.com> wrote:
*Hi,
Conference is working well before. Suddenly it is not working i have tested manually by calling 3002 conference room from Soft phone .But call get hangup directly with the ivr message BYE.
I have given "^(30\d{2})$" in default.xml . what is the error?
I have pasted the console logs in this path http://pastebin.freeswitch.org/6617
Try to correct where is error and what is the error????
-- Warm Regards, N.Baskar *
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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
participants (4)
-
Anthony Minessale
-
Baskar
-
Dan Le
-
Michael Jerris