search for: autofallthrough

...is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten => _X.,2,NoOp(DIALSTATUS=${DIALSTATUS}:HANGUPCAUSE=${HANGUPCAUSE}) ;exten => _X.,3,Hangup I also have autofallthrough=yes in the [general] section. The problem occurs when I dial an extension that returns busy. The Dial returns immediately and goes straight on to the NoOp, which confirms the busy status, but the system then waits 10 seconds before hanging up the calling Zap channel, even though autofallthrough is...

...IP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup channels and to use DeadAGI. I want it to not send the hangup to the provider at all. Taking out the Hangup line does not help nor does autofallthrough=no. I have posted my extensions.conf below. Thanks - [general] autofallthrough=yes context=default [default] ;exten => _.,1,Dial(SIP/003214093773@3213084999,70,t) exten => _.,1,AGI(mta_auth.agi,${EXTEN}) exten => _.,2,Hangup -- Brian Wilkins Software Engineer brian@hcc.net Heritage...

...t localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are no...

...(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) Thank you in advance....

...t-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 attachment was scru...

...e ~ 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, answer the line and then Dial...

...? ? ? ? ? ? ?IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ?; ? ? ? ? ? ? ?Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =><did>:<password>@69.59.234.67:5060/202 [vonage-out] username=<did> type=friend secret=<password> port=5061 nat=yes host=69.59.234.67 fromuser=<did> fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 context=from-pstn canreinvite=no Here is the CLI c...

...ing 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,133,1) exten => 9...

Hello! How do I handle calls when they reach a queue that has no members? Currently, the callers are thrown out, because of the autofallthrough. The message is app_queue.c:2094 queue_exec: Unable to join queue 'queue-name' == Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN' It seems that Queue() won't continue at a specific priority - like n+101 - if there are no members in the queue. Any ideas?...

...arco and the Realtime Extensions in my extensions.conf. The problem is that when I exit from my Marco, I should return to my calling context, which is default but the next step for it should be switch statement which will use realtime extension. Somehow I am getting the following error below with autofallthrough=yes : -- Executing NoOp("SIP/555-5dcf", "Channel is SIP/555-5dcf") in new stack == Auto fallthrough, channel 'SIP/555-5dcf' status is 'UNKNOWN' And the following error with autofallthrough=no : -- Executing NoOp("SIP/555-f121", "Channel is SI...

...ponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup To what this related? About my extensions.conf file, I set priorityjumpin = yes and I set autofallthrough = no (and I am sure it is not related to the problem with ResponseTimeout application). Any help? Regards Bilal __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

...ext=internal secret=6011 host=dynamic ;client on same LAN as Asterisk nat=no ;extension for IP phone [6012] type=friend context=internal secret=6012 host=dynamic ;client on same LAN as Asterisk nat=no ;====================== extensions.conf [general] static=yes writeprotect=yes clearglobalvars=no autofallthrough=yes [vosp-incoming] exten => s,1,Dial(SIP/6011) exten => s,n,Hangup [internal] exten => 6011,1,Dial(SIP/6011) exten => 6011,n,Hangup exten => 6012,1,Dial(SIP/6012) exten => 6012,n,Hangup include => outgoing [outgoing] ;Route calls starting with 0 to VOSP exten => _0.,1,Di...

...ensions in my > extensions.conf. The problem is that when I exit from my Marco, I > should return to my calling context, which is default but the next > step for it should be switch statement which will use realtime > extension. Somehow I am getting the following error below with > autofallthrough=yes : > > -- Executing NoOp("SIP/555-5dcf", "Channel is SIP/555-5dcf") in new stack > == Auto fallthrough, channel 'SIP/555-5dcf' status is 'UNKNOWN' > > And the following error with autofallthrough=no : > > -- Executing NoOp("SIP/...

...0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV lookups on outbound callscontext=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15minexpiry=14;rtautoclear=no;autofallthrough=yes register =>16194077214:<<password>@69.59.234.67:5060/202 [authentication][3000]type = friendcontext = defaultusername = 3000host = dynamicmailbox = 3000dtmfmode = rfc2833[3001]type = friendcontext = defaultusername = 3001host = dynamicmailbox = 3001dtmfmode = rfc2833 [3002]type = fr...

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten => _89859716,1,Dial(SIP/202) [macro-sipmail] exten => s,1,Verbose(1,Extension ${ARG1}) ;line req to pick up ext if it's not reg. exten => s,n,Dial(SIP/${ARG1},30) exten => s,n,GotoIf($[&q...

...te voip.eutelia.it:5060 <username> 8585 Registered It seems that all is all right. But, when I try to call a number from kphone (on datile3), I listen a message which say: the number you are dialing it does not exists. The .conf files: 1) extension.conf: [general] static=yes autofallthrough=yes clearglobalvars=no priorityjumping=no [eutelia] include => out_eutelia exten=>_XXXXXXXXXX,1,Dial(SIP/100@out_eutelia,20) exten => _XXXXXXXXXX,2,Hangup 2) sip.conf: [general] context=eutelia realm=voip.eutelia.it port=5060 bindaddr=0.0.0.0...

...friend | host=dynamic | username=ekiga1 | secret=6641ekiga1 | disallow=all | allow=ulaw | [ekiga2] | type=friend | host=dynamic | username=ekiga2 | secret=6641ekiga2 | disallow=all | allow=ulaw `---- And the significant lines of my extensions.conf ,---- | [general] | static=yes | writeprotect=no | autofallthrough=yes | clearglobalvars=no | priorityjumping=no | [globals] | CONSOLE=Console/dsp | IAXINFO=guest | TRUNK=Zap/g2 | TRUNKMSD=1 | exten => 555,1,Dial(SIP/ekiga1) | exten => 556,1,Dial(SIP/ekiga2) `---- Monroux Philippe (French Reunion island) -- - Pourquoi les capitalistes devraient installer L...

...dialplan=unknown group=1 callerid=asreceived ;Sangoma A101 port 1 [slot:3 bus:2 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel => 1-15,17-31 Extensions ----------------------------------------------------------------------- [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [dundi-e164-local] switch => DUNDi/e164 [local] ignorepat => 9 include => default include =&...

...> > context=incoming > > disallow=all > > allow=ulaw > > allow=alaw > > allow=g729 > > allow=g723 > > externip=72.220.28.226 > > localnet=192.168.0.0 > > nat=yes > > maxexpiry=15 > > minexpiry=14 > > ;rtautoclear=no > > ;autofallthrough=yes > > > > register =><did>:<password>@69.59.234.67:5060/202 > > > > [vonage-out] > > username=<did> > > type=friend > > secret=<password> > > port=5061 > > nat=yes > > host=69.59.234.67 > > fromuser=<d...

...pr 27 08:16:00 WARNING[3497]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' mi configuration files are this: extensions.conf: [general] static=yes writeprotect=yes ;autofallthrough=yes ;clearglobalvars=no ;priorityjumping=no [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =...