[local_stream://moh] already broadcasting...broadcast aborted
Full log and trace here: http://pastebin.freeswitch.org/11062 Pretty standard situation. User calls another user (same profile) and tries to place the call on hold (RFC 3264/sendonly). FS places call on hold and tries to start music but ends with: [local_stream://moh] already broadcasting...broadcast aborted ... and we don't get music. Any ideas why? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
I noticed you are sending the call to a socket... what did you do to the call in the socket? /b On Nov 11, 2009, at 3:19 PM, Kristian Kielhofner wrote:
Full log and trace here:
http://pastebin.freeswitch.org/11062
Pretty standard situation. User calls another user (same profile) and tries to place the call on hold (RFC 3264/sendonly). FS places call on hold and tries to start music but ends with:
[local_stream://moh] already broadcasting...broadcast aborted
... and we don't get music. Any ideas why?
Thanks!
From the trace:
# 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody@192.168.4.192 Command Execute set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec) # EXECUTE sofia/pjsip/nobody@192.168.4.192 set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec) # 2009-11-11 11:23:58.909804 [DEBUG] mod_dptools.c:766 sofia/pjsip/nobody@192.168.4.192 SET [sip_h_X-voalte-call-id]=[9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody@192.168.4.192 Command Execute ring_ready() # EXECUTE sofia/pjsip/nobody@192.168.4.192 ring_ready() # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540 sofia/pjsip/nobody@192.168.4.192 Command Execute bridge({originate_timeout=30,bypass_media=false,origination_caller_id_number=1001,origination_caller_id_name=Danielle Reed}sofia/voalte/huttoj@192.168.4.180) # EXECUTE sofia/pjsip/nobody@192.168.4.192 bridge({originate_timeout=30,bypass_media=false,origination_caller_id_number=1001,origination_caller_id_name=Danielle Reed}sofia/voalte/huttoj@192.168.4.180) # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 0 = [originate_timeout=30] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 1 = [bypass_media=false] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 2 = [origination_caller_id_number=1001] # 2009-11-11 11:23:58.909804 [DEBUG] switch_ivr_originate.c:1357 variable string 3 = [origination_caller_id_name=Danielle Reed] # 2009-11-11 11:23:58.909804 [NOTICE] switch_channel.c:613 New Channel sofia/pjsip/huttoj@192.168.4.180 [66e0041f-feff-49c1-baf8-0e5aa1ae99fa] 1) Set a SIP header (this is being removed, actually). 2) Indicate ring_ready (it's a long story). 3) Execute bridge to call the other device. On Wed, Nov 11, 2009 at 4:52 PM, Brian West <brian@freeswitch.org> wrote:
I noticed you are sending the call to a socket... what did you do to the call in the socket? /b
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64. On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
From the trace:
..snip.. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
dont execute bridge that way, your bridge itself is the other thing already broadcasting. api uuid_transfer <uuid of chan> bridge:sofia/myprofile/foo@bar.com inline if you want to do more after the bridge set the variable park_after_bridge=true to make it go back to idle On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64.
On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
From the trace:
..snip..
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
Finally got a chance to test this, the results are the same. Why am I getting this? Is it because I'm executing ring_ready before I attempt the bridge? Is it because I'm using a socket? On Wed, Nov 11, 2009 at 10:14 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
dont execute bridge that way, your bridge itself is the other thing already broadcasting.
api uuid_transfer <uuid of chan> bridge:sofia/myprofile/foo@bar.com inline
if you want to do more after the bridge set the variable park_after_bridge=true to make it go back to idle
On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64.
On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
From the trace:
..snip..
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
_______________________________________________ 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
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
results cant possibly be the same there is not even any broadcast involved in uuid_transfer ? you need to attach a console trace with debug log up On Fri, Nov 20, 2009 at 10:41 AM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Finally got a chance to test this, the results are the same.
Why am I getting this? Is it because I'm executing ring_ready before I attempt the bridge? Is it because I'm using a socket?
On Wed, Nov 11, 2009 at 10:14 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
dont execute bridge that way, your bridge itself is the other thing already broadcasting.
api uuid_transfer <uuid of chan> bridge:sofia/myprofile/foo@bar.cominline
if you want to do more after the bridge set the variable park_after_bridge=true to make it go back to idle
On Wed, Nov 11, 2009 at 4:33 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64.
On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote:
From the trace:
..snip..
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
-- 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
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
As always, you are correct. The scenario now is: - If the caller places the callee on hold, the callee will get hold music - If the callee places the caller on hold, the caller will not get hold music I've uploaded a fresh pastebin here: http://pastebin.freeswitch.org/11356 On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
results cant possibly be the same there is not even any broadcast involved in uuid_transfer ?
you need to attach a console trace with debug log up
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
I decided to just change the code so its more elegant to handle recursive broadcasting so you can try again and see if that helps. On Wed, Dec 2, 2009 at 10:35 AM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
As always, you are correct.
The scenario now is:
- If the caller places the callee on hold, the callee will get hold music - If the callee places the caller on hold, the caller will not get hold music
I've uploaded a fresh pastebin here:
http://pastebin.freeswitch.org/11356
On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
results cant possibly be the same there is not even any broadcast involved in uuid_transfer ?
you need to attach a console trace with debug log up
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
Tony, Thanks for that but now it appears that the call just gets hung up on when the caller takes the callee off hold. Debug here: http://pastebin.freeswitch.org/11359 Thanks again! On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I decided to just change the code so its more elegant to handle recursive broadcasting so you can try again and see if that helps.
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
I am not sure what you are sending over the socket but you have a queued hangup being processed on line 640 of your pastebin are you executing any commands with a ! character in it by any chance or executing the hangup app on purpose? On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Tony,
Thanks for that but now it appears that the call just gets hung up on when the caller takes the callee off hold. Debug here:
http://pastebin.freeswitch.org/11359
Thanks again!
On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I decided to just change the code so its more elegant to handle recursive broadcasting so you can try again and see if that helps.
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
Try trunk again On Wed, Dec 2, 2009 at 5:33 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I am not sure what you are sending over the socket but you have a queued hangup being processed on line 640 of your pastebin are you executing any commands with a ! character in it by any chance or executing the hangup app on purpose?
On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Tony,
Thanks for that but now it appears that the call just gets hung up on when the caller takes the callee off hold. Debug here:
http://pastebin.freeswitch.org/11359
Thanks again!
On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I decided to just change the code so its more elegant to handle recursive broadcasting so you can try again and see if that helps.
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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
Tony, The call no longer hangs up but we still only get hold music in one direction - if the callee places the caller on hold there is no music. PB here: http://pastebin.freeswitch.org/11378 This was on rev 15773. Thanks again Tony! On Thu, Dec 3, 2009 at 10:56 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try trunk again
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
participants (3)
-
Anthony Minessale
-
Brian West
-
Kristian Kielhofner