search for: ast_pbx_run

Hello, Can anyone help with the output shown below? It?s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 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' Jul 7 18:42:32 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into inval...

...ew stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time Aug 6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'ip2pri' Aug 6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'ip2pri' Aug 6 11:52:40 DEBUG[81926]: chan_h323.c:1179 cleanup_connection: Cleaning up our mess My configs are: h323.conf:...

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

...- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at incoming,s,1 still failed so falling back to context 'default' Dec 13 18:12:32 WARNING[2499]: pbx.c:1878 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' Dec 13 18:12:42 NOTICE[2500]: chan_zap.c:5361 ss_thread: Got event 2 (Ring/Answered)... ==...

...;80 90 a3> LLC = default HLC = <91 81> AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=**some_number**,CID=**some_number**,CIP=0x10,CONTROLLER=0x1) Jun 24 22:19:49 WARNING[1086696368]: pbx.c:1819 ast_pbx_run: Channel 'CAPI[contr1/**some_number**]/0' sent into invalid extension 's' in context 'default', but no invalid handler -- CAPI Hangingup > activehangingup -- started pbx on channel (callgroup=0)! -- INFO_IND ID=002 #0x011e LEN=0023 Controller/P...

...IAXINFO=guest ; IAXtel username/password TRUNK=Zap/1 ; Trunk interface [local] include => default [default] exten => 203755343,1,Dial(Zap/g1/203755343) The error I get is Sep 17 14:18:20 WARNING[1185515456]: pbx.c:1839 ast_pbx_run: Timeout, but no rule 't' in context 'default' -- Executing Dial("H323/ip$80.247.147.146:3852/19860", "Zap/g1/203755343") in new stack -- Couldn't call g1/203755343 -- Hungup 'Zap/1-1' h323 --> Asterisk ---> h323 works ok -------...

...TICE[29298]: chan_zap.c:5458 ss_thread: Got event 2 (Ring/Ans wered)... == Starting Zap/1-1 at incoming,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at incoming,s,1 still failed so falling back to context ' default' Nov 28 08:55:09 WARNING[29298]: pbx.c:1963 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' My extensions.conf file looks like this: [incoming] s,1,Dial(Zap/2,15) s,2,Voicemail(u10) s,3,Hangup My voicemail.conf looks like this: [def...

...5.67.76.30:5060> 2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1255 pbx_extension_helper: Launching 'Congestion' 2004-08-09 17:36:29 DEBUG[245775]: channel.c:652 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/65.67.76.30-0814e4f0' 2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1827 ast_pbx_run: Spawn extension (bogon-calls,5462000,1) exited non-zero on 'SIP/65.67.76.30-0814e4f0' 2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1255 pbx_extension_helper: Launching 'Congestion' 2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1947 ast_pbx_run: Spawn extension (bogon-calls,h,1) exit...

...============ == Starting H323/ip$209.237.227.185:46373/20161 at incoming,18005551212,1 failed so falling back to exten 's' == Starting H323/ip$209.237.227.185:46373/20161 at incoming,s,1 still failed so falling back to context 'default' May 7 14:19:18 WARNING[30649]: pbx.c:1889 ast_pbx_run: Channel 'H323/ip$209.237.227.185:46373/20161' sent into invalid extension 's' in context 'default', but no invalid handler ================================================================= The remote-test server sends the h323 call perfectly, but the receiving server isn...

...error *CLI> 0:18.190 H225 Caller:8112978 H225 Received connect PDU. 0:18.288 H245:810b388 H245 Read error: Bad file descriptor 0:18.318 H323 Cleaner H323 Connection ip$localhost/25430 terminated. WARNING[23576]: File pbx.c, Line 1702 (ast_pbx_run): Timeout, but no rule 't' in context 'voip-h323' 0:28.313 H225 Answer:80fc988 H225 Read error (4): Interrupted system call 0:28.349 H323 Cleaner H323 Connection ip$172.18.1.184:2127/5227 terminated. This is what i have in oh323.conf conte...

I get the following message when I dial 74#. Does anyone have any ideas on what might be going on? If I don't require numbers to be terminated with # everything works as expected, but you have to wait for the digit timeout, of course. MESSEGE: DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got something to jump out with ('#')! -- Invalid extension '#' in context 'speeddial' on SIP/2111-076f CONFIG: [default] exten => 74#,1,Goto(speeddial,s,1) [speeddial] exten => s,1,Background(${MYSOUNDS}/enterspeedcode) exten => 1#,1,Background(${MY...

...-- Executing Dial("SIP/6200-70bb", "SIP/6203| 20") in new stack -- Called 6203 -- Got SIP response 486 "Busy Here" back from 10.1.2.116 -- SIP/6203-d4f0 is busy == Everyone is busy/congested at this time Sep 30 15:19:26 WARNING[52240]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'default' This is what I have in the dialplan [macro-oneline] exten => s,1,Dial(${ARG1}, 20) [default] exten => 6200,1,Macro(oneline,SIP/6200) exten => 6201,1,Macro(oneline,SIP/6201) Best Regards Meng Kim...

...7 13:45:19 WARNING[1032209]: channel.c:1901 ast_request: No channel type registered for 'H323' Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742 dial_exec: Unable to create channel of type 'H323' == Everyone is busy/congested at this time Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'default' [general] static=yes writeprotect=no ;Trunk=Modem/g1 [default] exten => 2004,1,NoOp( call for ${EXTEN}) exten => 2004,2,Dial(SIP/${EXTEN},10,tr) exten => 2004,3,Congestion exten => 2005,1,NoOp( call for ${EXTEN}) ext...

...2 (Critical Request) == No one is available to answer at this time Dec 7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 40dedd1535853f17250b4d0854e35c17@200.75.243.237 for seqno 102 (Non-critical Request) Dec 7 17:05:26 WARNING[-176354384]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'sip' -- Executing Dial("SIP/erick2-db3b", "SIP/erick1") in new stack -- Called erick1 Dec 7 17:05:56 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 5184f4e66bc98ae361f5f5f512858a9f@...

...ot;, '\0' <repeats 38 times> tmp3 = "\e[1;35;40mzap/g1/00551138856342|120|rtS(10883)\e[0;37;40m\000accountcode:102190|UserID:3456|src:33225075|srcip:217.157.177.77|ConnectPrice:30|PeakPrice:60|RateID:55|CustomerID:30001|DestNameInt:Brazil_S?o"... #7 0x08078c74 in ast_pbx_run (c=0x43a45fa8) at pbx.c:1879 digit = 0 '\0' exten = '\0' <repeats 255 times> pos = 0 waittime = 1180700108 res = 0 #8 0x080804e1 in pbx_thread (data=0xfffffffc) at pbx.c:2102 No locals. #9 0x40033f60 in pthread_start_thread () fro...

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten => 0,3,ResponseTimeout,10 exten => 0,4,waitexten any help appreciated. thanks liaan _________________________________________...

.../2000-6850", "ZAP/g1/3487024125") in new stack -- Called g1/3487024125 -- Zap/1-1 is making progress passing it to SIP/2000-6850 -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' == No one is available to answer at this time Aug 23 17:03:07 WARNING[6876]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'home' -- Executing NoOp("SIP/2000-6396", "03487024125") in new stack -- Executing Dial("SIP/2000-6396", "ZAP/g1/03487024125") in new stack -- Called g1/03487024125 -- Zap/1-1 is making progress passing...

....0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13 22:47:07 NOTICE[8767]: rtp.c:435 ast_rtp_read: RTP: Received packet with bad UDP checksum Dec 13 22:47:07 WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no rule 't' in context 'internal' linux*CLI> Someone could help me to fix the following problens?? fraternatly, Cleyverson Pereira Costa ------------------------------------------------ Phone #: 27+9922-0111 Skype: cleyverson MSN: cleyverson@hotmail.com ------...

...the console: -- Executing VoiceMail("Zap/10-1", "u200") in new stack -- Playing '/data/asterisk/var/spool/asterisk/voicemail/default/ 200/unavail' (language 'en') -- Redirecting Zap/10-1 to fax extension May 25 01:19:34 WARNING[17629]: pbx.c:2412 ast_pbx_run: Timeout, but no rule 't' in context 'answer-extension' -- Hungup 'Zap/10-1' It seems that Asterisk once entered in a Macro is unable to jump to the fax extension and gave me a timeout (which I do not handle in my dialplan). If I change the Wait(3) into Wait(0) t...

...lowed the examples for the conf files to the letter. I can call the pots extensions OK from IAX clients, SIP clients and from the incoming X100P cards. But, if I pick up the handset to make a call all I get is the engaged tone and the following message. May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1' sent into invalid extension 's' in context 'default' but no invalid handler. If I am reading my configs then shouldn't they be going to the internal context? Do I need to set-up pots extensions somewhere like IAX & Sip extensions? ==============...