search for: app_stack

...7 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'app_voicemail_plain.so': /usr/lib64/asterisk/modules/app_voicemail_plain.so: undefined symbol: ast_smdi_mwi_set [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'app_stack.so': /usr/lib64/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'app_speech_utils.so': /usr/lib64/asterisk/modules/app_speech_utils.so: undefined symbol: ast_speech_dtmf [Sep...

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

...IAX2/testuser,210") [Mar 15 16:14:06] DEBUG[6060]: pbx.c:2679 pbx_extension_helper: Launching 'Set' -- Executing [s at macro-call-ext:1] Set("SIP/testuser-remote-082b11f8", "LOCAL(arg1)=SIP/testuser&IAX2/testuser") in new stack [Mar 15 16:14:06] ERROR[6060]: app_stack.c:370 local_write: Tried to set LOCAL(arg1), but we aren't within a Gosub routine .. which means I'm not able to access these arguments inside the Macro :( FWIW, I've defined this Macro in extensions.ael (not .conf). Any ideas?? Thanks in advance! - Gonzalo -------------- next part...

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) Here are the segments of the ael: --------------------------------------- ... Dial(${MARKCELL},30,tgU(screen)); ... macro screen() { Wait(0.5); Read(ACCEPT,followme/options,1,,...

Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 loader.c: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/ap...

I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would

Hello, This is a update on the current status of Asterisk in Debian. Apologies for the really long mail, it is targetted both to users and maintainers :) I'm Ccing asterisk-users as a one-time thing; users that are interested can subscribe to our list[1] for updates to prevent noise on a non-Debian list. Please Cc pkg-voip-maintainer on replies. 1:

I need to use app_macro, but it seems to be absent from asterisk 16.30.1 Is there a workaround? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230710/8b3814ad/attachment.html>

...n callerid, I'd expect that to go to the invalid extension - "i" - and Return(). But look what happens: -- Executing [2136398447 at teliax-in:4] Gosub("IAX2/poseidon-15117", "set-callerid-name,02136990505,1") in new stack [Dec 25 13:06:32] ERROR[26483]: app_stack.c:286 gosub_exec: Attempt to reach a non-existent destination for gosub: (Context:set-callerid-name, Extension:02136990505, Priority:0) == Spawn extension (teliax-in, 2136398447, 5) exited non-zero on 'IAX2/johnfbeatty-15117' -- Hungup 'IAX2/poseidon-15117' -- Hungup...

...:51] NOTICE[9031]: pbx.c:4119 pbx_extension_helper: No such label 'stdexten' in extension '1' in context 'home' [Apr 3 18:20:51] WARNING[9031]: pbx.c:10174 pbx_parseable_goto: Priority 'stdexten' must be a number > 0, or valid label [Apr 3 18:20:51] ERROR[9031]: app_stack.c:411 gosub_exec: Gosub address is invalid: '1,stdexten(SIP/phone1)' I've googled and pretty much tried all forms of the syntax but I've yet to make it work. For instance I tried not including stdexten and calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work...

...eadaudio_callback: Failed to write frame Or: -- Executing [s at check-fax-reception:2] Wait("SIP/7005-00000291", "0.5") in new stack == Spawn extension (check-fax-reception, s, 2) exited non-zero on 'SIP/7005-00000291' [Aug 13 16:36:32] NOTICE[24309][C-00000470]: app_stack.c:1031 gosub_run: SIP/7005-00000291 Abnormal 'Gosub(check-fax-reception,s,1(SIP/7012-00000290,7005))' exit. Popping routine return locations. My understanding is : when a fax is detected and sent a fax extension, callee channel is in a state "close to termination" which is not...

...queue when reloading (Reported by Italo Rossi) * ASTERISK-22846 - testsuite: masquerade super test fails on all branches (still) (Reported by Matt Jordan) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between PGSQL database...

...queue when reloading (Reported by Italo Rossi) * ASTERISK-22846 - testsuite: masquerade super test fails on all branches (still) (Reported by Matt Jordan) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by Krzysztof Chmielewski) * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between PGSQL database...

...s on all branches (still) (Reported by Matt Jordan) * ASTERISK-23545 - Confbridge talker detection settings configuration load bug (Reported by John Knott) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-23616 - Big memory leak in logger.c (Reported by ibercom) * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS (Reported by Sebastian Wiedenroth) * ASTERISK-23550 - Newer sound sets don't show up in menu...

...s on all branches (still) (Reported by Matt Jordan) * ASTERISK-23545 - Confbridge talker detection settings configuration load bug (Reported by John Knott) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-23616 - Big memory leak in logger.c (Reported by ibercom) * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS (Reported by Sebastian Wiedenroth) * ASTERISK-23550 - Newer sound sets don't show up in menu...

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

...ttings configuration load bug (Reported by John Knott) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-22904 - bridges: lock the bridge when creating bridge snapshots (Reported by Matt Jordan) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-23616 - Big memory leak in logger.c (Reported by ibercom) * ASTERISK-23588 - ARI: Crash when unsubscribing from bridge (Reported by Matt Jordan) * ASTERISK-23502 - Channel variable SIPREFERTOHDR not being set d...

...ttings configuration load bug (Reported by John Knott) * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think (Reported by Walter Doekes) * ASTERISK-22904 - bridges: lock the bridge when creating bridge snapshots (Reported by Matt Jordan) * ASTERISK-23620 - Code path in app_stack fails to unlock list (Reported by Bradley Watkins) * ASTERISK-23616 - Big memory leak in logger.c (Reported by ibercom) * ASTERISK-23588 - ARI: Crash when unsubscribing from bridge (Reported by Matt Jordan) * ASTERISK-23502 - Channel variable SIPREFERTOHDR not being set d...

..._helper: Launching 'Gosub' [Aug 31 14:41:15] DEBUG[6964]: res_xmpp.c:3494 xmpp_client_receive: XML parsing successful -- Executing [s at xmpp-incoming:1] Gosub("Message/ast_msg_queue", "xmpp-incoming,message,1") in new stack [Aug 31 14:41:15] DEBUG[6954][C-00000000]: app_stack.c:578 gosub_exec: Channel Message/ast_msg_queue has no datastore, so we're allocating one. [Aug 31 14:41:15] DEBUG[6954][C-00000000]: pbx.c:6065 __ast_pbx_run: Extension message, priority 0 returned normally even though call was hung up The exact specifics of the debug after priority 1 varies...

...diosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming call local Contact header - extconfig: Allow explicit DB result set ordering to be disabled. - rest-api: Run make...