search for: pbx_extension_helper

...code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing Answer("SIP/200-d83a", "") in new stack Aug 20 18:57:45 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-d83a' -- Executing Answer("SIP/200-81f6", "") in new stack Aug 20 18:57:59 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No app...

...caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk stasis.c:1490 publish_msg() # 6: [0x59588e] asterisk stasis_channels.c:796 ast_channel_publish_snapshot() # 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec() # 8: [0x52e039] asterisk pbx.c:2989 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: [0...

I have added a sip user in sip.conf. user name is 819,context is c819. and I have added the follows rows in extension.conf. like [c819] exten => 1,1,Answer exten => 1,2,SetVal(voicemail=${exten}) exten => 1,3,Dial(SIP/${voicemail}) It appear a error message(pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 2)) when i dial 1 from 819. The version of asterisk is 1.0.3 Please help me. Thank a lot. Bill Chen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attach...

...to? Thanks, Tom DEBUG: Connected to Asterisk 1.0.7 currently running on dev1 (pid = 26799) Verbosity is at least 10 -- Executing Goto("SIP/4.68.250.152-08129478", "validatenumber|s|1") in new stack -- Goto (validatenumber,s,1) Jun 28 01:01:02 WARNING[26800]: pbx.c:1291 pbx_extension_helper: No application 'agi' for extension (validatenumber, s, 1) == Spawn extension (validatenumber, s, 1) exited non-zero on 'SIP/4.68.250.152-08129478' EXTENSIONS.CONF: [validatenumber] exten => s,1,agi(test.agi) exten => s,2,HangUp

Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -------------- next part -------------- An HTML attachment was scrubbed...

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming, 5147771111, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extens...

...risk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-00000013] pbx.c: NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE! vs. VERBOSE[7609][C-00000013]: pbx.c:2925 in pbx_extension_helper: NOTICE[3042]: chan_sip.c:30421 in sip_poke_noanswer: Peer '8884' is now UNREACHABLE! specifically, the messages coming from syslog have extra debugging information (which I am not interested in): pbx.c: chan_sip.c: vs . pbx.c:2925 in pbx_extension_helper: chan_sip.c:30421 in...

....21 Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 3359 (check_user): Setting NAT on RTP to 0 Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 2899 (build_route): build_route: Contact hop: 5010 <sip:5010@62.212.12.21> Jun 5 01:55:33 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'Answer' Jun 5 01:55:33 DEBUG[1236360496]: File chan_sip.c, Line 934 (sip_answer): sip_answer(SIP/5010-d3c4) Jun 5 01:55:33 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'BackGround' Jun 5 01:55:33 DEBUG[1236360496]: File channel.c, Line 1...

Hi All: I have experienced some big problems on an asterisk production server under 1.2.18: First of all, a very rare message like this... No application Macro ??? -- Saved useragent "Linksys/SPA922-5.1.7" for peer 1363 Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No application 'Macro' for extension (pbx-incoming, 1133, 1) == Spawn extension (pbx-incoming, 1133, 1) exited non-zero on 'SIP/1210-081aa708' Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No application 'Macro' for extension (pbx-incoming, h, 1) == Spa...

....conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco (below) I get : Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Which is correct, meaning the context declaration is not being respected. ------ dial-peer voice 101 voip destination-pattern 10. session protocol sipv2 session target ipv4:10.0.0.133 dtmf-relay rtp-nte codec g711ulaw no vad ------- My bad...

...mingfd=-1) [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6397 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 9 08:42:42] DEBUG[18512]: channel.c:1026 channel_find_locked: Avoiding initial deadlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: pbx.c:1795 pbx_extension_helper: Launching 'Wait' [May 9 08:42:42] DEBUG[18512]: devicestate.c:287 do_state_change: Changing state for SIP/128.91.56.38 - state 2 (In use) [May 9 08:42:42] DEBUG[18518]: chan_sip.c:4393 find_call: = Found Their Call ID: 9CB723DE-FD6111DB-9FBF9C45-B28951D9@128.91.56.38 Their Tag 479EE6...

...c etc etc .. -- Starting simple switch on 'DAHDI/3-1' -- Executing [s at in:1] Set("DAHDI/3-1", "CDR(userfield)=23XXXXX6") in new stack -- Executing [s at in:2] Goto("DAHDI/3-1", "in2") in new stack [Nov 14 16:45:51] NOTICE[29607]: pbx.c:4522 pbx_extension_helper: No such label 'in2' in extension 's' in context 'in' [Nov 14 16:45:51] WARNING[29607]: pbx.c:10888 pbx_parseable_goto: Priority 'in2' must be a number > 0, or valid label == Spawn extension (in, s, 2) exited non-zero on 'DAHDI/3-1' -- Hanging up on...

...st_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319 #11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577 #12 0xb7097dc5 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 0x...

...leutt 5) (utt.send.wave.client wholeutt))) I have placed the above text before the last line which is (provide 'festival). Below is the debug log shown on asterisk console : [Aug 4 17:50:11] > Channel SIP/gafachi1a-00000000 was answered. [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Answer' [Aug 4 17:50:11] -- Executing [s at connect-to-me:1] Answer("SIP/gafachi1a-00000000", "") in new stack [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'SayDigits' [Aug 4 17:50:11] -- Executing [s at connect...

...me. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Set' -- Executing [s at tbsendfax:1] Set("SIP/vltb-sbc01-00000000", "timestarted=20100512224709") in new stack [May 12 22:47:09] DEBUG[22725]: pbx.c:3692 pbx_extension_helper: Launching 'Answer' -- Executing [s at tbsendfax:2] Answer(&quo...

...able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'from-internal' May 26 07:35:23 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'from-internal' my extension.conf is :- [globals] VM_PREFIX = * RINGTIMER = 15 REGTIME = 7:55-17:05 REGDAYS = mon-fri RECORDEXTEN = "&q...

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

...ple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display : Jul 25 09:48:09 WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No application 'VoiceMailMain ' for extension (home, 22999, 1) Anybody knows why? Ciao and thank you! Mauro Zanin

Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asteri...

pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in extension 'PHONE NUMBER' in context 'phones' [Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto: Priority 'outgoing|PHONE NUMBER' must be a number > 0, or valid label PHONE NUMBER = the number I called....