search for: writeprotect

Is there anyway to unset the extensions.conf definition of writeprotect=yes while in the CLI interface (or by other mechanism) to enable the dialplan save command? I accidentally overwrote my extensions.conf but still have a running copy of asterisk with the old dial plan running in memory. while it would not be difficult for me to rebuild what I lost - it would be...

...-------------------- My current setup: On computer 192.168.10.50 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0 context=default allow=gsm register=>comp1:mysecret@192.168.10.51 [comp2] type=friend username=comp2 secret=bigsecret host=192.168.10.51 **extensions.conf*** [general] static=yes writeprotect=no [default] exten=>_221,1,Dial,SIP/comp2 -================= On computer 192.168.10.51 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0 context=default allow=gsm register=>comp2:bigsecret@192.168.10.50 [comp1] type=friend username=comp1 secret=mysecret host=192.168.10.50 **extensions.c...

...on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is forwarded the call is still ringing? My current /etc/asterisk/extensions.conf file looks like this: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [pstn] exten => s,1,NoOp(Caller ID is ${CALLERID(num)}) exten => s,2,Dial(Zap/1,15,g2) exten => s,n,Congestion [local] ignorepat => 9 exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) exten => _9.,n,Congestion exten => 11,1,Dial(Zap/1,20,rt) Th...

...060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here [globals] [inside] exten => 77,1,voicemailmain [other] exten => 88,1,Playback(demo-congrats) Next, I have an x-lite phone set up as Display name: 40 Username: 40 Authorization user: 40 Domain/Realm: 69.240.152.95 SIP Proxy: 69.240.15...

...ation zapata.conf [channels] context=default signalling=r2 r2country=ar group=1 callgroup=1 pickupgroup=1 transfer=yes echocancel=yes callprogress=yes immediate=yes channel => 1-15,17-31 /etc/zaptel span=1,0,0,cas,hdb3 cas=1-15:1001 cas=17-31:1001 Extentions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/1 ; Trunk interface [local] include => default [default] exten => 203755343,1,D...

...rld Dialup) type=user context=sip [xlite1] "Transmit Silence"=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid="Jane Smith" <5678> host=dynamic allow=ulaw allow=alaw (extensions.conf) [general] static=yes writeprotect=no [analog] include=>test include=>local [sip] include=>test include=>local [test] 611,1,echo_test [local] exten => 1237,1,Dial(SIP/xlite1,10,t)

...nel 'SIP/test-3a26' status is 'CONGESTION' #/etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 #/etc/asterisk/zapata.conf: [channels] switchtype=national context=default signalling=pri_cpe group=1 channel => 1-23 #/etc/asterisk/extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes [default] exten => 123,1,Answer() exten => 123,2,Playback(hello-world) exten => 123,3,Hangup() exten => _9NXXXXXX,1,Dial(Zap/g1) Any ideas? Thank you in advance, your help is greatly appreciated. -Dan -------------- next part -------------- An HTML...

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically calls the right mailbox using the ${CALLERIDNUM} variable in the current context (var ${CONTEXT}). exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) include => i...

...The Dialplans are simple...caller machine just plays a 3 minutes gsm and loops, and the callee machine dilaplan launches an AGI that plays some gsm, records 20 secs of the call and hangups the call. Server#1 (caller) extensions.conf -------------------------------------- [general] static=yes writeprotect=no [inicio] exten => s,1,PlayBack(laxana) exten => s,2,Goto(s,1) exten => t,1,hangup exten => i,1,hangup exten => o,1,hangup exten => h,1,hangup -------------------------------------- Server#2 (callee) extensions.conf -------------------------------------- [general] static=yes w...

...rname=hurricane to 172.22.0.50 and 404 error on the sip phone. here are my extension.conf and iax.conf for both servers. for hurricane: ------------------------------- extensions.conf ------------------------------- ; [general] ; ; ; XXX Not yet implemented XXX ; static=yes ; ; if static=yes and writeprotect=no, you can save dialplan by ; CLI command 'save dialplan' too ; writeprotect=no ; ; The "Globals" category contains global variables that can be referenced ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable ; ${${VARIABLE}} or ${text${VARIABLE}} or...

...he message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Vo...

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executi...

...quot;Jefferson Carvalho" <1260> host=dynamic nat=no canreinvite=no allow=gsm ; [1262] type=friend context=sip username=1262 secret=1262 qualify=300 callerid="Ialle" <1262> host=dynamic nat=no canreinvite=no allow=gsm ; *>> My extensions.conf * [general] static=yes writeprotect=no [globals] CONSOLE => Console/dsp IAXINFO => guest TRUNK => Zap/g2 TRUNKMSD => 1 [sip] exten => 1260,1,Dial(SIP/1260,20) exten => 1261,1,Dial(SIP/1261,20) exten => 1262,1,Dial(SIP/1262,20) Best Regards, -Jefferson Carvalho IT Analist Credishop S/A Teresina-PI-Brasil...

...verlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include => customer exten => h,1,Hangup exten => i,1,Congestion exten => i,2,Hangup [agnosco] include => local-extensions include => customer_ivr include => incoming [customer_ivr] include => local-extensions exten =>...

...~ username = username ~ password=password ~ context=montr?al ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf ------------------------------------------(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montr?al] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:password@SERVER2.DOMAIN.COM/51412345678@montr?al) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answ...

...:40 DEBUG[81926]: chan_h323.c:1179 cleanup_connection: Cleaning up our mess My configs are: h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 gatekeeper = DISABLE context=ip2pri [ip2pri] ; is this needed? type=user context=ip2pri extensions.conf: [general] static=yes writeprotect=yes [globals] [ip2pri] exten => _9.,1,Dial(Zap/1/${EXTEN:0}) ; i must send the 9 to the PRI... [default] zapata.conf: [channels] context=pri2ip switchtype=euroisdn signalling=pri_net ; pri_net is ok group=1 channel => 1-15,17-31 Some help? Thanks, HoraPe --- Horacio J. Pe?a horape...

...cific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no ;OEM exten => _12125551212,1,Goto(OEM,s,1) [OEM] exten => s,1,Answer() exten => s,n,Set(CALLERID(name)=OEM - ${CALLERID(number)}) exten => s,n,Background(Outsource) exten => s,n,WaitExten(10) exten => s,n,Goto(inside,1...

...nPhone type=friend defaultip=193.25.30.223 username=223 context=default [register] ..... [codecs] codec=G711A frames=20 codec=G711U frames=20 codec=GSM0610 frames=4 /////////////////////////////////////////////// /////////////////////////////////////////////// Extension.conf [general] static=yes writeprotect=no [default] ;Xlite exten => 224,1,SetLanguage(de) exten => 224,2,Dial(SIP/xlite1,10) exten => 224,3,Voicemail(u224) exten => 224,102,Voicemail(b224) exten => 224,103,Hangup ;Openphone (H.323) exten => 223,1,SetLanguage(de) exten => 223,2,Dial(OH323/193.25.30.223,15)...

...' 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 ${EXTEN}) exten => 2005,2,Dial(SIP/${EXTEN},10,tr) exten => 2005,3,Congestion exten => _61XXXX,1,Dial...

...=192.168.3.60 bandwidth=low permit=192.168.3.205 ;register => user0:secret0@192.168.3.205 [site1] type=friend host=192.168.3.205 username=user1 secret=secret1 auth=md5 context=incoming trunk=yes qualify=1000 disallow=all allow=ilbc voip:/etc/asterisk# more extensions.conf [general] static=yes writeprotect=no [globals] ; Global Variables ; Internal SIP Phone Numbers PHONE1=SIP/2001 ; Other Site Authentication SITE1=IAX2/user1:secret1@site1 ; MACRO SECTION [macro-callextension] exten => s,1,Dial(${ARG1}) exten => s,2,Hangup exten => s,102,Playtones(busy) exten => s,103,Wait,30 exten =&g...