search for: __ast_pbx_run

...989 pbx_extension_helper() # 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension() #10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec() #11: [0x53b599] asterisk pbx_app.c:493 pbx_exec() #12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper() #13: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run() #14: [0x53184b] asterisk pbx.c:4669 decrease_call_count() #15: [inlined] asterisk pbx.c:4702 pbx_thread() #16: [0x5b8329] asterisk utils.c:1576 dummy_start() #17: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack() #18: [0x7f62bd0fe8dd] libc.so.6 :0 clone() [2023-11-08 18:14:13]...

[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone...

...n=0xb22bf9a8, data=Variable "data" is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffffffc, data=0x7fffffff) at app_dial.c:1680 #7 0x08090bad in pbx_extension_helper (c=0xb22bf9a8, con=Variable "con" is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xb22bf9a8) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffffffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 5 (process 11504): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1...

...extension --> Caller hears extension ring on receiver --> Call goes through Caller (Night) --> Dials an extension --> Caller hears silence until vm picks up --> Leaves a voicemail... Significant albeit insanely stupid Asstricks message: 2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got something to jump out with ('2')! (Oooh how about creating errors we can figure out Digium!) Any thoughts -- ==================================================== J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 sil . infiltrated @ net http://www.inf...

...7dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619 #13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 "op05_x", exten=0xb659ff14 "00116", priority=1, label=0x0, callerid=0x0, action=0) at pbx.c:553 #14 0x0808efea in __ast_pbx_run (c=0xb659fcd0) at pbx.c:2227 #15 0x0808fcdf in pbx_thread (data=0xb7ed1900) at pbx.c:2514 #16 0xb7f7cb63 in start_thread () from /lib/tls/libpthread.so.0 #17 0xb7e7718a in clone () from /lib/tls/libc.so.6 another one: #0 0xb6ff38e2 in decodeMP3 () from /usr/lib/asterisk/modules/format_mp3.so #1...

...Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

...n=0x82496a8, data=0xb720f828) at app_queue.c:3601 #4 0x080c638d in pbx_extension_helper (c=0x82496a8, con=0x64, context=0x8249828 "my-queue", exten=0x8249878 "80", priority=7, label=0x0, callerid=0x821e5b0 "562390", action=136482640) at pbx.c:532 #5 0x080c7041 in __ast_pbx_run (c=0x82496a8) at pbx.c:2314 #6 0x080c7fd1 in pbx_thread (data=0x64) at pbx.c:2631 #7 0x080f7e99 in dummy_start (data=0x64) at utils.c:843 #8 0xb7f7f13d in pthread_start_thread () from /lib/libpthread.so.0 #9 0xb7ea81ba in clone () from /lib/libc.so.6 (gdb) bt full #0 0xb7e5a231 in strcasec...

...console shows that: *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 26 19:34:33 WARNING[3818]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Apr 26 19:34:38 NOTICE[3821]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... Apr 26 19:...

...ow can I setup my Chan Dongle recived calls in my Asterisk? I have to setup in dongle.conf ? Or in extensions.conf? And the code for recive I found this site http://asterisk-service.com/page/chan-dongle-use I have to To save Subscriber Number before? See the error log in my Asterisk pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-0100000000' sent to invalid extension but no invalid handler: context,exten,priority=URA,+5511965380290,1,Noop(),1 Thanks in advanced.

...gt; s,n,Goto(s,1) exten => s(notEmpty),n,Background(my/thank-you) exten => s,n,Wait(1) When I receive call and tries to enter the digits (86 lets say), it only accept just first digit and terminates even before considering second digit. Error message : WARNING[5726][C-0000000a]: pbx.c:6696 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'testmacro' Please suggest what might be wrong. Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail...

...7 oad:0878722291 with 's' extension P[ 1] MGMT: SSTATUS: L1_ACTIVATED == Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's' == Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to context 'default' Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2' sent into invalid extension 's' in context 'default', but no invalid handler 087822291 is the number i dial from, 0594643637 is the number that the asterisk server should respond to. in misdn.conf i created a kpn section like this: [kpn] ports=1ptmp...

...n=0x0, data=0x1) at app_dial.c:1601 #4 0x08090aed in pbx_extension_helper (c=0xa281820, con=0x0, context=0xa281970 "default", exten=0xa281a64 "2348053004990", priority=2, label=0x0, callerid=0xf46a40b0 "ss7/08053004990|60", action=0) at pbx.c:544 #5 0x08091db6 in __ast_pbx_run (c=0xa281820) at pbx.c:2218 #6 0x0809386c in pbx_thread (data=0x0) at pbx.c:2505 #7 0x00c161d5 in start_thread () from /lib/tls/libpthread.so.0 #8 0x00a972da in clone () from /lib/tls/libc.so.6 (gdb) bt full #0 ast_var_name (var=0x1) at chanvars.c:71 name = 0xffffffff <Address 0xffff...

...551212 at LegA-f707,2 == Starting Local/16505551212 at LegA-f707,1 at default,callback,1 failed so falling back to exten 's' == Starting Local/16505551212 at LegA-f707,1 at default,s,1 still failed so falling back to context 'default' [Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel 'Local/16505551212 at LegA-f707,1' sent into invalid extension 's' in context 'default', but no invalid handler Uhm, why? I have a default context with a callback extension. Of course I have no explicit priority 1 though... this is AEL2.... What's it complaini...

...; (language 'en') -- Executing Wait("SIP/9001-08fb34d0", "2") in new stack -- Executing MeetMe("SIP/9001-08fb34d0", "12345|p") in new stack -- Playing 'conf-getpin' (language 'en') Oct 26 18:52:47 WARNING[23516]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'from-sip' -- Hungup 'Zap/pseudo-656465881' Oct 26 18:53:35 WARNING[23485]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'from-sip' * / -- Thank...

...; (language 'en') -- Executing Wait("SIP/9001-08fb34d0", "2") in new stack -- Executing MeetMe("SIP/9001-08fb34d0", "12345|p") in new stack -- Playing 'conf-getpin' (language 'en') Oct 26 18:52:47 WARNING[23516]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'from-sip' -- Hungup 'Zap/pseudo-656465881' Oct 26 18:53:35 WARNING[23485]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'from-sip' */ -- Thanks...

...P> SIP/2.0 -- Executing [s at default:1] Answer("SIP/sip.jmg.se-081dd730", "") in new stack [2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol 01@<ASTERISK_IP> SIP/2.0 [2007-10-17 19:23:56] WARNING[10123]: pbx.c:2505 __ast_pbx_run: Timeout, but no rule 't' in context 'default' Last line already fixed by adding such a rule - just Hangup() for now...

Hello, I'm having troubles while using the "Goto" function in a realtime extension. Here is the error message : -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1") -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my e...

...ee in the context below: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup But i have the following error when trying to dial 111: [Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel 'SIP/michofr-093833e0' sent into invalid extension '111' in context ' custom-recordme', but no invalid handler Any help? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attach...

...The issue is that the call is matching the test context but as soon as it execute the GoTo tag I got the following error in the log: Executing Goto("SIP/gw-in.dddd.net-b7803718", "On-net|028945551|1") -- Goto (On-net,028945551,1) [May 14 20:38:13] WARNING[8462]: pbx.c:2470 __ast_pbx_run: Channel 'SIP/gw-in.dddd.net-b7803718' sent into invalid extension '028945551' in context 'On-net', but no invalid handler It seems that the GoTo is not working well here...Can someone help me in that please? Regards -------------- next part -------------- An HTML attachme...