search for: stdexten

...eting); exten => *,1,Directory(default) exten => 205,1,Wait(2) exten => 205,2,Record(/tmp/asterisk-recording:gsm) exten => 205,3,Wait(2) exten => 205,4,Playback(/tmp/asterisk-recording) exten => 205,5,Wait(2) exten => 205,6,Hangup [internal] exten => 101,1,Macro(stdexten,SIP/101) exten => 102,1,Macro(stdexten,SIP/102) exten => 103,1,Macro(stdexten,SIP/103) exten => 123,1,Macro(stdexten,SIP/123) exten => 124,1,Macro(stdexten,SIP/124) exten => 125,1,Macro(stdexten,SIP/125) exten => 126,1,Macro(stdexten,SIP/126) exten => 127,1,Macro(stdexte...

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten written in anything else but extensions.conf (AEL, LUA, etc, being these not able to define an arbitrary priority) will not work. The only way to workaround this...

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489", "__DYNAMIC_FEATURES=") in new stack -- Executing [s at macro-stdexten:2] GotoIf("IAX2/6223-10489", "0?5:3") in new stack -- Goto (macro-stdexten,s,...

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten => s,102,Voicemail2(b${A...

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-std...

...mbine in and out together. Any suggestions? console log (Test A) - Outside line calling in ============================================== -- Accepting call from '298741000' to '98553166' on channel 0/10, span 1 -- Executing [98553166 at default:1] Macro("Zap/10-1", "stdexten|5166|5166") in new stack -- Executing [s at macro-stdexten:1] SetMusicOnHold("Zap/10-1", "cpwr") in new stack -- Executing [s at macro-stdexten:2] Set("Zap/10-1", "DYNAMIC_FEATURES=automon") in new stack -- Executing [s at macro-stdexten:3] Dial("Za...

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten => 1,1,Gosub(${EXTEN},stdexten(SIP/phone1)) ==============...

...have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i have: exten => 8029,1,Macro(stdexten,8029) and in stdexten macro: exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-NOANSWER,2,Goto(default,s,1) exten => s-BUSY,1,Voicemail(b${ARG1}) exten => s-BUSY,2,Goto(default,s,1) exten => _s-.,1,Goto(s-NOANSWER,1) exten => a,1,Voice...

...record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log: -- Accepting call from '298741000' to '98553166' on channel 0/10, span 1 -- Executing [98553166 at default:1] Macro("Zap/10-1", "stdexten|5166|5166") in new stack -- Executing [s at macro-stdexten:1] SetMusicOnHold("Zap/10-1", "cpwr") in new stack -- Executing [s at macro-stdexten:2] Set("Zap/10-1", "DYNAMIC_FEATURES=automon") in new stack -- Executing [s at macro-stdexten:3] Dial("Za...

...> requested prefs = (g729), > actual format = g729, > host prefs = (g729), > priority = mine -- Executing Set("IAX2/Bris-IAX-1", "LDAPNUMBERCALLED=xxxx 2015") in new stack -- Executing Macro("IAX2/Bris-IAX-1", "stdexten|2015|2015") in new stack -- Executing Dial("IAX2/Bris-IAX-1", "sip/2015|20|twh") in new stack -- Called 2015 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'IAX2/Bris-IAX-1' in macro 'stdexten' == Spawn extension (macro-stdexten...

...rror below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing Macro("SIP/508-226d", "stdexten|SIP/505") in new stack -- Executing Dial("SIP/508-226d", "|60") in new stack Feb 1 12:34:04 WARNING[2250]: app_dial.c:504 dial_exec: Dial argument takes format (technology1/number1&technology2/number2...|optional timeout) == Spawn extension (macro-stdexten, s, 1)...

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake?? [macro-std...

...t/20545/tmp/9mrzpn format: gsm, 0x33c7938 -- x=2, open writing: /var/spool/asterisk/voicemail/default/20545/tmp/9mrzpn format: wav, 0x33c0f18 [Oct 23 11:03:33] WARNING[8832][C-00000003]: file.c:830 ast_readaudio_callback: Failed to write frame -- User hung up == Spawn extension (default, stdexten-NOANSWER, 1) exited non-zero on 'SIP/20321-00000001' == Using SIP RTP CoS mark 5 -- Executing [20545 at default:1] Gosub("SIP/20321-00000002", "20545,stdexten(SIP/20545)") in new stack -- Executing [20545 at default:50000] NoOp("SIP/20321-00000002", &...

...oS mark 5 [Mar 26 12:11:04] -- Executing [5421047051 at restrito:1] Goto("SIP/pabx-e1-00000252", "interno,7051,1") in new stack [Mar 26 12:11:04] -- Goto (interno,7051,1) [Mar 26 12:11:04] -- Executing [7051 at interno:1] Macro("SIP/pabx-e1-00000252", "stdexten,7051,SIP/7051") in new stack [Mar 26 12:11:04] -- Executing [s at macro-stdexten:1] NoOp("SIP/pabx-e1-00000252", "STDEXTEN: Arg1 = 7051 Arg2 = SIP/7051 Arg3 = ") in new stack [Mar 26 12:11:04] -- Executing [s at macro-stdexten:2] Dial("SIP/pabx-e1-00000252&qu...

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

...14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/411@default-d873,1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3001 Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Macro("Local/413@default-2959,2", "stdexten|413|SIP/413") in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Set("Local/413@default-2959,2", "DYNAMIC_FEATURES=automon") in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Dial("Local/413@default-2959,2", "SIP/413...

...sip_buddies sippeers => mysql,asterisk,sip_buddies voicemail => mysql,asterisk,sip_buddies extensions => mysql,asterisk,extensions queues => mysql,asterisk,sip_buddies queue_members => mysql,asterisk,sip_buddies extensions.conf: [office] include => demo ;exten => 333,1, Macro(stdexten,333,SIP/333) ;exten => user1, 1, Goto(333|1) ;exten => 222,1, Macro(stdexten,222,SIP/222) ;exten => user1, 1, Goto(222|1) ;exten => 201,1, Macro(stdexten,201,SIP/201) ;exten => user2, 1, Goto(201|1) ;exten => 202,1, Macro(stdexten,202,SIP/202) ;exten => user3, 1, Goto(202|1) sw...

...12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/411@default-eaff,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3001 Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Macro("Local/413@default-0edc,2", "stdexten|413|SIP/413") in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Set("Local/413@default-0edc,2", "DYNAMIC_FEATURES=automon") in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Dial("Local/413@default-0edc,2", "SIP/413...

...] and both don't work If I want to call the mobile phone (6201) with a Twinkle soft phone (6000) I get following message in the CLI-window from Asterisk: > == Using SIP RTP CoS mark 5 > -- Executing [6201 at DLPN_DialPlan1:1] Macro("SIP/6000-00000013", > "stdexten,6201,SIP/6201") in new stack > -- Executing [s at macro-stdexten:1] Set("SIP/6000-00000013", > "__DYNAMIC_FEATURES=") in new stack > * [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror: > ast_yyerror(): syntax error: syntax error, unexpec...

...d hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten => 187,hint,SIP/zyx2000 exten => 187,1,Macro(stdexten,447,SIP/zyx2000,187) the entries in the table for realtime config look like this: SELECT context, exten AS ext, priority AS prio, app, appdata FROM extensions WHERE exten IN (183,187); +----------+-----+------+...