I have converted the normal Park application and I can only alert you about the syntax change. I suspect also in the ParkAndAnnounce command, the parameters are ordered completely different.
Leandro 2014-01-30 Anders Larsson <[email protected]>: > Hi > > I'm trying to get the rebuilt parking functionality to work in Asterisk > 12.0.0. > > In Asterisk 11.6.0 I managed to get a call to get parked by adding a > dynamic feature in features.conf for the DMTF sequence *# which called a > macro in extensions.conf, which then runned the ParkAndAnnounce > application, and the call got parked. > > The syntax for ParkAndAnnounce I used was this (I don't want any > announcement to be played): > > exten => s,n,ParkAndAnnounce(,3600,SIP/100) > > > In the new Asterisk-version, the ParkAndAnnounce application gets called, > but the call isn't parked. > > The only error I can see in the messages file is a DEBUG entry saying that > the channel "failed to join Bridge", like this: > > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge_channel.c:1994 > bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: > 0x16e3768(SIP/vpn-sbc-00000001) failed to join Bridge > > > Anyone else that has tried to convert old parking functionality into > Asterisk 12.0.0 ? > > > > features.conf: > > parkswitch => *#,callee/caller,Macro(parkswitch) > > > extensions.conf: > > [default] > .... > > include => parkedcalls > > [macro-parkswitch] > exten => s,1,ParkAndAnnounce(,,PARKED,SIP/100) > > > messages: > > [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2847 > create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at x.x.x.x:9530 > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4050 __ast_read: DTMF > begin '*' received on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4061 __ast_read: DTMF > begin passthrough '*' on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2165 > ast_rtp_update_source: Setting the marker bit due to a source update > [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2847 > create_dtmf_frame: Creating END DTMF Frame: 42 (*), at x.x.x.x:9530 > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:3964 __ast_read: DTMF > end '*' received on SIP/at-tcty-ssw-00000000, duration 240 ms > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4005 __ast_read: DTMF > end accepted with begin '*' on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4034 __ast_read: DTMF > end passthrough '*' on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:00] DEBUG[7114][C-00000000]: bridge_channel.c:1174 > bridge_channel_feature: DTMF feature string on > 0x7f6b8c10f998(SIP/at-tcty-ssw-00000000) is now '*' > [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2847 > create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at x.x.x.x:9530 > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4050 __ast_read: DTMF > begin '#' received on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4054 __ast_read: DTMF > begin ignored '#' on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:01] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2847 > create_dtmf_frame: Creating END DTMF Frame: 35 (#), at x.x.x.x:9530 > [Jan 30 21:00:01] DTMF[7114][C-00000000]: channel.c:3964 __ast_read: DTMF > end '#' received on SIP/at-tcty-ssw-00000000, duration 230 ms > [Jan 30 21:00:01] DTMF[7114][C-00000000]: channel.c:4034 __ast_read: DTMF > end passthrough '#' on SIP/at-tcty-ssw-00000000 > [Jan 30 21:00:01] DEBUG[7114][C-00000000]: bridge_channel.c:1174 > bridge_channel_feature: DTMF feature string on > 0x7f6b8c10f998(SIP/at-tcty-ssw-00000000) is now '*#' > [Jan 30 21:00:01] DEBUG[7114][C-00000000]: bridge_channel.c:1185 > bridge_channel_feature: DTMF feature hook 0x7f6b8c1d9480 matched DTMF > string '*#' on 0x7f6b8c10f998(SIP/ssw-00000000) > [Jan 30 21:00:01] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2165 > ast_rtp_update_source: Setting the marker bit due to a source update > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: res_rtp_asterisk.c:2165 > ast_rtp_update_source: Setting the marker bit due to a source update > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: app.c:305 ast_app_exec_macro: > SIP/vpn-sbc-00000001 Original location: default,,1 > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: pbx.c:4875 > pbx_extension_helper: Launching 'ParkAndAnnounce' > -- Executing [s@macro-parkswitch:1] > ParkAndAnnounce("SIP/vpn-sbc-00000001", ",,PARKED,SIP/100") in new stack > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 > find_best_technology: Bridge technology softmix does not have any > capabilities we want. > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 > find_best_technology: Bridge technology simple_bridge does not have any > capabilities we want. > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 > find_best_technology: Bridge technology native_rtp does not have any > capabilities we want. > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:505 > find_best_technology: Chose bridge technology holding_bridge > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:771 bridge_base_init: > Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: calling holding_bridge > technology constructor > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:779 bridge_base_init: > Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: calling holding_bridge > technology start > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge_roles.c:272 > setup_bridge_role: Set role 'holding_participant' > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge_channel.c:1977 > bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: > 0x16e3768(SIP/vpn-sbc-00000001) is joining > *[Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge_channel.c:1994 > bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: > 0x16e3768(SIP/vpn-sbc-00000001) failed to join Bridge* > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: app_macro.c:428 _macro_exec: > Spawn extension (macro-parkswitch,s,1) exited non-zero on > 'SIP/vpn-sbc-00000001' in macro 'parkswitch' > == Spawn extension (macro-parkswitch, s, 1) exited non-zero on > 'SIP/vpn-sbc-00000001' in macro 'parkswitch' > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: app.c:308 ast_app_exec_macro: > Macro exited with status -1 > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: app.c:322 ast_app_exec_macro: > SIP/vpn-sbc-00000001 Ending location: default,,1 > [Jan 30 21:00:01] DEBUG[7118][C-00000000]: res_rtp_asterisk.c:2165 > ast_rtp_update_source: Setting the marker bit due to a source update > [Jan 30 21:00:01] DEBUG[7119]: taskprocessor.c:484 > tps_taskprocessor_destroy: destroying taskprocessor > '423a711c-02c7-4b54-ab39-33e6c64e32c3' > [Jan 30 21:00:01] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:3284 > ast_rtcp_read: Got RTCP report of 76 bytes > [Jan 30 21:00:02] DEBUG[7118][C-00000000]: res_rtp_asterisk.c:3284 > ast_rtcp_read: Got RTCP report of 76 bytes > [Jan 30 21:00:05] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:3284 > ast_rtcp_read: Got RTCP report of 76 bytes > [Jan 30 21:00:07] DEBUG[7118][C-00000000]: res_rtp_asterisk.c:3284 > ast_rtcp_read: Got RTCP report of 76 bytes > [Jan 30 21:00:10] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:3284 > ast_rtcp_read: Got RTCP report of 76 bytes > > -- Anders > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
