search for: _7xx

...low. Julian. #### extensions.conf #### [general] static=yes writeprotect=yes [default] exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,BackGround(tt-monkeysintro) exten => s,6,BackGround(tt-monkeys) exten => _7XX,1,Dial(zap/1/${EXTEN}|5m) exten => _7XX,2,Voicemail(u${EXTEN}) exten => _7XX,3,Hangup exten => _7XX,103,Voicemail(b${EXTEN}) exten => _7XX,104,Hangup exten => #,1,Playback(demo-thanks) exten => #,2,Hangup exten => t,1,Goto(#,1) exten => i,1,Playback(invalid) exten => i,...

...ce quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as _7XXX and all users can then be registered with the server and get allocated the their individual numbers by the server.(till now i define the numbers with callerid field). i need to do this so that i can add 15+ users without having to add each individually. what would be the entry inthe iax.conf for t...

...layed as soon as they answer the phone - they do not have to enter "#" Extensions.conf: [from-zap] exten => 800,1,AgentCallBackLogin(|@callq) [from-sip] exten => 800,1,AgentCallBackLogin(|@callq) [callq] exten => 811,1,Dial(SIP/811,10) exten => 811,2,Hangup exten => _7XX,1,Dial(Zap/g2/${EXTEN},10) exten => _7XX,2,Hangup Ackcall=no is set in agents.conf (I've also tried ackcall=0) Any clues ? CVS head as of 10 minutes ago! Julian.

This possibly isn't 100% asterisk related, but I'd like some opinions/feedback... A customer wanted different ring-tones to differentiate external and internal calls. No biggie once I'd worked out that details - they have 100% GXP2000 phones, so adding in the relevant SIP header and altering the phones to suit seems like it's going to be a solution... But I started to look at

...30182849,2,Dial(SIP/bt-rlm,45,t) exten => 30182849,3,Voicemail,u550 exten => 30182849,103,Voicemail,b550 [te405p-in] exten => _2XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _2XX,2,Hangup exten => _73816592XX,1,Dial(Zap/g4/${EXTEN:-3},60,r) exten => _73816592XX,2,Hangup exten => _7XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _7XX,2,Hangup exten => _1XXX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _1XXX,2,Hangup include => sip include => parkedcalls include => te405p-outgoing include => transfer-record [te405p-ext] exten => s,1,SetMusicOnHold(random) exten =&gt...

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

Hello, I know not too long ago I saw /something/ _somewhere_ about an adjustment to call parking that allowed blind transfers from SIP phones to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to

...hanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: *************************************** extensions.conf >>>> *************************************** [default] ..... ..... [outgoing] exten=>_7XX,1,Goto(voip-h323|${EXTEN}|1) exten=>_*XX,1,Goto(servicios|${EXTEN}|1) exten=>_XXXXXXXXX,1,Dial(Zap/1/${EXTEN}|30) exten=>_XXXXXXXXX,2,Playback(invalid) exten=>_XXXXXXXXX,3,Hungup() exten=>_X,1,Playback(invalid) exten=>_X,2,Hungup exten=>_XX,1,Playback(invalid) exten=>_XX,2...

...ons.conf [dialstring] exten => i,1,Playback(invalid) exten => i,2,Hangup exten => t,1,Hangup [te405p-frombp250] include => to-sip include => te405p-outtelstra [te405p-tobp250] exten => _2XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _4XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _7XX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _1XXX,1,Dial(Zap/g4/${EXTEN},60,r) exten => _73816592XX,1,Dial(Zap/g4/${EXTEN:7},60,r) [te405p-intelstra] exten => s,1,SetMusicOnHold(record) exten => s,2,Dial(SIP/bt-pavilion,45,t) exten => s,4,VoiceMail,u500 exten => s,5,Hangup exten =&gt...

I am newbie trying to setup about 12 Polycom Ip500's on an asterisk server. I am working on my extensions.conf and am trying to make it so that all my extensions can dial each other. My extensions are number 720, 721, 722, 723 ..etc in my from-sip context I began doing entries such as: exten => 720,1,Dial(SIP/720,20) exten => 720,2,Voicemail(u720) exten =>

...?3) exten => h,3,SetGlobalVar(EMERGENCY=0) [to-sip] #include extensions_sip.conf [queue_admin_ext] exten => _2XX,1,Dial(ZAP/g4/1111${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding)) exten => _3XX,1,Dial(SIP/${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding)) exten => _7XX,1,Dial(ZAP/g4/1111${EXTEN},20,mA(lm_features/Queues/pls_say_thankyou_holding)) [lm1_functions] #include extensions_lm1a.conf #include extensions_js_play.conf #include extensions_night_switch.conf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.co...