search for: u200

...s: [zap] exten => s,1,Answer exten => s,2,ChanIsAvail(${DERRICK}) exten => s,3,SetVar,"EVERYONE=${DERRICK}" exten => s,4,ChanIsAvail(${DON}) exten => s,5,SetVar,"EVERYONE=${EVERYONE}&${DON}" ... exten => s,10,Dial(${EVERYONE},20) exten => s,11,Voicemail,u200 exten => s,111,Voicemail,b200 exten => s,13,Hangup exten => t,1,Hangup Which caused Asterisk not to ring any phones or goto voicemail. Here is the section of our extensions.conf : DERRICK=SIP/202 DON=SIP/203 THOMAS=SIP/204 JAMES=SIP/205 BRUCE=SIP/206 EVERYONE=${DERRICK}&${DON}&...

...exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten => 300,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 300,4,SetCIDNum(0${CALLERIDNUM}) exten => 300,5,dial(${TRUNK}c/2831385,20,r) exten => 300,6,SetCIDNum(${NewCaller}) exten => 300,7,voicemail2(u200@default) exten => 300,101,voicemail2(b200@default) exten => 300,102,hangup Regardless of what (and where) I have tried to implement this, I just cant get it to work properly. Does anyone have some tips on this or a nicer follow-me type of script? Kerry Garrison http://www.geekgazette.com...

...ait, 2- what happens if 5s is still not enough another time. So Asterisk recognized that a fax was being received while executing the macro stdfwd3iax-notransfer (this extension simply check And I saw the following message in 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-ext...

..._." matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; ; ;exten => _.,1,Congestion [default] ;Extension 200 Cordless Phone exten => 200,1,Dial(SIP/200,20) exten => 200,2,Voicemail(u200) exten => 200,102,Voicemail(b200) exten => 200,103,Hangup ;Extension 203 Office Phone exten => 203,1,Dial(SIP/203,20) exten => 203,2,Voicemail(u200) exten => 203,102,Voicemail(b200) exten => 203,103,Hangup ;Extension 303 Office Phone exten => 303,1,Dial(SIP/303,20) exten =&gt...

..._." matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; ; ;exten => _.,1,Congestion [default] ;Extension 200 Cordless Phone exten => 200,1,Dial(SIP/200,20) exten => 200,2,Voicemail(u200) exten => 200,102,Voicemail(b200) exten => 200,103,Hangup ;Extension 203 Office Phone exten => 203,1,Dial(SIP/203,20) exten => 203,2,Voicemail(u200) exten => 203,102,Voicemail(b200) exten => 203,103,Hangup ;Extension 303 Office Phone exten => 303,1,Dial(SIP/303,20) exten =&gt...

..._." matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; ; ;exten => _.,1,Congestion [default] ;Extension 200 Cordless Phone exten => 200,1,Dial(SIP/200,20) exten => 200,2,Voicemail(u200) exten => 200,102,Voicemail(b200) exten => 200,103,Hangup ;Extension 203 Office Phone exten => 203,1,Dial(SIP/203,20) exten => 203,2,Voicemail(u200) exten => 203,102,Voicemail(b200) exten => 203,103,Hangup ;Extension 303 Office Phone exten => 303,1,Dial(SIP/303,20) exten =&gt...

...s callprogress=yes busydetect=yes busycount=6 echocancelwhenbridged=no echotraining=800 rxgain=5.5 group=0 channel=2 channel=3 group=1 usecallerid=yes channel=4 ---extensions-custom.conf--- [custom-incoming] exten => s,1,Dial(Zap/g0/0408xxxxxx&Sip/200,30,t) exten => s,104,Voicemail(u200) exten => s,105,Hangup()

Hello, I have a weird problem between Asterisk 13.3 and a Yeastar U200 pbx over IAX trunk. Should I call from Yeastar to my asterisk 13.3 the call goes through without issues. Should I call from asterisk 13.3 to Yeastar I can hear a ring tone however the yeastar does not show any activities. On the yeastar I initiated a debug command iax2 set debug peer "m...

Hi I am trying to make asterisk load config from database, so far I get the sip, extension working, but voicemail seems still looking for config file, not from the database. the extconfig.conf looks like ... sipusers => mysql,asterisk,sip_buddies sippeers => mysql,asterisk,sip_buddies extensions => mysql,asterisk,extensions_table voicemail => mysql,asterisk,voicemail_users .. the

...-- Executing NoOp("SIP/208-f55b", "Using CallerID "abc" <208>") in new stack -- Executing Goto("SIP/208-f55b", "s-NOANSWER|1") in new stack -- Goto (macro-vm,s-NOANSWER,1) -- Executing VoiceMail("SIP/208-f55b", "u200@default") in new stack -- Playing 'vm-theperson' (language 'en') == Spawn extension (macro-vm, s-NOANSWER, 1) exited non-zero on 'SIP/208-f55b' in macro 'vm' == Spawn extension (macro-vm, s-NOANSWER, 1) exited non-zero on 'SIP/208-f55b' in macro...

Dear list, following problem, i have some users, who are supressing their callerid. This setting is adjusted at the sip phone. So if these guys are calling internal persons nobody sees the callerid. I am looking for the following resolution: User has set his phone to anonymous, user calls somebody internal, Asterisk initials a lookup on the channel and generates a new callerid for the

...,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1) exten => s,3,Dial(SIP/201,25,tr) exten => s,4,DIal(SIP/211&SIP/202&SIP/203&SIP/209&SIP/211&SIP/212&SIP/213&SIP/214,15,tr) exten => s,5,Background(/etc/asterisk/day) exten => s,6,Wait(3) exten => s,7,Voicemail(u200@fakepbxname) exten => s,8,Hangup [main-night-aa] exten => s,1,Answer exten => s,2,Background(/etc/asterisk/night) exten => s,3,Voicemail(u201@fakepbxname) exten => s,4,Hangup exten => 1,1,Directory(fakepbxname,internal,l) exten => 00,1,VoicemailMain(s@fakepbxname) When in nig...

>1- Call comes in without callerid >2- AGI script answers line >3- AGI script asks to record name >4- Park the call and get the parked extension number >5- Ring all the phones in the house (exec Dial) >6- If phone is picked up, play recorded name >7- Wait for DTMF to accept or decline call >8- If accepted, bridge parked call and current call. Mike, I am wanting this

....100.200.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 190.100.200.19:5060 -- SIP/101-8b54 is busy == Everyone is busy at this time -- Executing Wait("SIP/200-409e", "2") in new stack -- Executing VoiceMail("SIP/200-409e", "u200") in new stack We're at 190.100.200.1 port 18090 Answering with preferred capability 1024 Answering with preferred capability 8 Answering with preferred capability 256 Answering with preferred capability 2 Answering with preferred capability 1 Answering with preferred capability 4 Answerin...