search for: pbx_config

...e. If the operator hangs up, all is well, and all the affected channels are cleared. Any tips to this? Busydetect is NO in zapata.conf for other reasons (calls being inadvertently dropped by asterisk). My Dialplan: pbxMobile*CLI> show dialplan [ Context 'default' created by 'pbx_config' ] Include => 'radiance' [pbx_config] Ignore pattern => '9' [ Context 'radiance' created by 'pbx_config' ] '9' => 1. Background(radiancedirectory) [pbx_config] 2. DigitTimeout(3) [pbx_config]...

...wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1. *Set(DB(blacklist/${blacknr})=1)* [pbx_config] 2. Playback(num-was-successfully) [pbx_config] 3. Playback(added) [pbx_config] 4. Wait(1)...

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

...can find and the closest I get tells me to use the script to create a mailbox, but there is no script. The second problem I am having is that my dialplan is not working how it used to, for reasons I cant quite put my fingure on; here is the dial plan: [ Context 'default' created by 'pbx_config' ] 's' => 1. Goto(well|3000|1) [pbx_config]...

...gup() [pasvalide] exten => _.,1,Answer() exten => _.,2,Playback(invalid) exten => _.,3,Playback(goodbye) exten => _.,4,Hangup() -- When I on asterisk's CLI try 'show dialplan' command this is what I get: *CLI> show dialplan [ Context 'pasvalide' created by 'pbx_config' ] '_.' => 1. Answer() [pbx_config] 2. Playback(invalid) [pbx_config] 3. Playback(goodbye) [pbx_config] 4. Hangup()...

..."SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by 'pbx_config' ] '0001' => -1. Dial() [pbx_config] 1. Dial(SIP/mvickers) [pbx_config] '0002' => -1. Dial() [pbx_config] 1. Dial(SIP/jgb) [pbx_config] '0003' => -1. Dial() [pbx_config] 1....

...f asterisk -vvvv -c -U asterisk, and the relevant output of show dialplan. Note that the sip calls come in on extension 666. Thanks much in advance. Ryan ======================================================================= show dialplan: [ Context 'in-from-7869101' created by 'pbx_config' ] '1' => 1. Dial(SIP/lesnet_peer/1218348xxxx|30|ptT) [pbx_config] 2. VoiceMail(200 at testingvm) [pbx_config] 3. Playback(vm-goodbye) [pbx_config] 4. Hangup()...

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config] 104. Macro(dia...

...1,1,Goto(extension_is_known,s,1) ; Allow users the ability to get Directory listing (user must be in voicemail.conf) exten => 2,1,Directory,default|internal_extensions here is the output of my show dialplan start: pbx01*CLI> show dialplan start [ Context 'start' created by 'pbx_config' ] '0' => 1. Goto(s|1) [pbx_config] '1' => 1. Goto(extension_is_known|s|1) [pbx_config] '2' => 1. Directory(default|internal_extensions) [pbx_config] 'i' =>...

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

...to get it working? I was led to believe that you would need the 't' at the end of the dial string to enable the called party to transfer the call about, is this correct? My dial plan for the relevant contexts looks a little like this: [ Context 'local-extensions' created by 'pbx_config' ] '0' => 1. Goto(2000|1) [pbx_config] '2001' => 1. Dial(SIP/2001|20|tr) [pbx_config] 2. Voicemail(u2001) [pbx_config] 102. Voicemail(b2001)...

...r>ww${EXTEN:1}) exten => _90027[1-3].,2,Congestion ; Canada - All Phones exten => _9001.,1,Dial(Zap/1/<access-number>ww${EXTEN:1}) exten => _9001.,2,Congestion ===== extensions.conf - end ===== ===== show dialplan - start ===== [ Context 'international' created by 'pbx_config' ] '_9001.' => 1. Dial(Zap/1/<access-number>ww${EXTEN:1}) [pbx_config] 2. Congestion() [pbx_config] '_90027[1-3].' => 1. Dial(Zap/1/<access-number>ww${EXTEN:1}) [pbx_config]...

...t was incorrect exten => _X.,7,Wait,2 exten => _X.,8,playback(fatfingers) exten => _X.,9,Wait,2 exten => _X.,10,playback(fatfingers) exten => _X.,11,hangup ;exten => h,1,playback(goodbye) and a lookie from the prompt: show dialplan last [ Context 'last' created by 'pbx_config' ] '#' => 1. hangup() [pbx_config] 'h' => 1. hangup() [pbx_config] 'i' => 1. hangup() [pbx_config] 't' =>...

...ple trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context 'outgoing' created by 'pbx_config' ] '123' => 1. Answer() [pbx_config] 2. AGI(/usr/local/share/examples/asterisk/agi/agi- test.agi) [pbx_config] 3. Hangup() [pbx_config] '_X.' => 1. Dial(SIP/sipprovider/${EXTEN}||L[3...

...he level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [ Context 'macro-stdexten' created by 'pbx_config' ] 'a' => 1. VoicemailMain(${ARG1}) [pbx_config] 's' => 1. Dial(${ARG2}|30) [pbx_config] 2. Goto(s-${DIALSTATUS}|1) [pbx_config] 's-BUSY' => 1....

Can someone refresh my memory how to backlist caller ID in asterisk 1.8? I had it working in ver. 1.4 but in 1.8 it changed. -- Joseph

...l from 'from-twilio' (UDP:mysillyApp.pstn.twilio.com:5060) to extension '+17775551212' rejected because extension not found in context 'from-twilio-remove-plus'. ip-xxxx*CLI> dialplan show from-twilio-remove-plus [ Context 'from-twilio-remove-plus' created by 'pbx_config' ] '_[+1]XXXXXXXXXX' => 1. GotoIf($["${CALLERID(num):0:2}" != "+1"]?noplusatstart) [pbx_config] 2. Log(WARNING,TWILIO) [pbx_config] 3. Set(CALLERID(num)=${CALLERID(num):1}) [pbx_config] [noplusatstart] 4. Goto(from-...

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2. Answer() [pbx_config] 3. Background(welcome) [pbx_config] 4. Background(and)...

I was wondering why *foo at default should match '_*[0-9a-zA-Z].*0.' in 1.6.13. Who is making the parse error, * or me? CLI> dialplan show *foo at default '_*[0-9a-zA-Z].*0.' => 1. NoOp(${EXTEN}) [pbx_config] 2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config] 4. Set(CDR(accountcode)=${accountcode}) [pbx_config] 7. ResetCDR() [pbx_config] 8. ... -- Daniel Tryba

On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by 'pbx_config' ] 's' => 1. SendFAX(${FaxFile}.tif) [pbx_config...