search for: exten

...;t do a proper routing, when I was trying out parked calls with key #, the phone which is connected to BB was constantly trying to connect to AA. Here is the Dialplan configuration: Asterisk server number one (AA) All outgoing calls need a 5555 prefix in front of the number [buster] ; btc trunk exten => _9.,1,DIAL(SIP/btctrunk/${EXTEN}) exten => _9.,n,Hangup() ; soft ; caller-id SOFT ;exten => _X.,1,Set(CALLERID(num)=015555${CALLERID(num)}) exten => _X.,1,Set(CALLERID(num)=0158631${CALLERID(num)}) exten => _X.,n,DIAL(SIP/softnet/555501${EXTEN}) exten => _X.,n,Hangup() ; popr...

...have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter what I set the ring strategy to. So I set an extension up to do a dynamic login for them, this worked out for about a day or so and it has started doing the same thing. Plus it drops calls sometimes too. I don't have a dropped call problem anywhere else in the company so I am assuming it is the queue. Can't find anything strange (to me) in...

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf Jeff Sorry for t...

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to voicemail... Also, the most annoying is when I try to place an outside call... At that point, if the call is to a regular phone number, I'll get a message aski...

...th the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring Dial Zap/g5/9399||T 00:07:58 (None) Obviously, this is a big problem for us... Below are my zapata.conf, zaptel.conf and extensions.conf: -------------------------- zapata.conf -------------------------- [channels] usecallerid=yes hidecallerid=no echocancel=yes musiconhold=service busydetect=yes ;callprogress=yes busycount=3 flash=20 rxflash=40 transfer=yes threewaycalling=yes ;rxgain=100% ;txgain=1.0 ;relaxdtmf=yes ;...

...I can call SOME of the numbers (like 337xxxx, and 998xxxx, and 323xxxx).. but when I try to dial say like 601xxxx I get a 404.. any thoughts, I can't see any difference in the config. Also, I seem to be able to dial any number that starts with a 9.. such as 977, 990, 903.. [outbound-local] ;exten => _NXXXXXX,1,Macro(dialout-default,${EXTEN}) ;exten => _NXXNXXXXXX,1,Macro(dialout-default,${EXTEN}) exten => _220NXXX,1,Macro(dialout-default,${EXTEN}) exten => _244NXXX,1,Macro(dialout-default,${EXTEN}) exten => _279NXXX,1,Macro(dialout-default,${EXTEN}) exten => _318NXXX,1,Mac...

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk is 1.6.2.11. Below is the extensions.conf file. [globals] QPHONE0=SIP/10 QPHONE1=SIP/11 QPHONE2=SIP/12 QPHONE3=SIP/13 QPHONE4=SIP/14 QPHONE5=SIP...

...____submenu Dial 1 for product a support Dial 2 for pdoduct b support Dial 3 for product c support My problem is that if you choose option one in the second menu it loop back to the first menu. I don't know how to handle this, and I'm sure it can be done. here is the section of extensions.conf that deals with it exten => 6666,1,Wait,2 ; Allow for PRI to grab info in facility exten => 6666,2,SetCallerID(Toll Free No Cpub) exten => 6666,3,BackGround(greeting) exten => 6666,4,BackGround(mainmenu) exten => 6666,5,Wait,5 exten => 6666,6,Queue(tech) exten => 1...

...ve a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to the outside world, every 2-5 minutes randomly it drops all active calls with the following error on console. == Spawn extension (te405p-intelstra, 38166483, 2) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/2-1' == Spawn extension (te405p-frombp250, 00402270...

...ystem, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the entensions on the ericsson or local sip phones no problems, if someone on a sip phone calls an extension on the ericsson it goes straight through no pause. If someone on the ericsson system dials a sip phone it takes close to 3 full seconds before the sip phone rings, it takes that long just to get to the asterisk box, although its not the ericsson phone system that is the problem, if I dump a st...

...5 nat=1 host=dynamic dtmfmode=rfc2833 canreinvite=yes reinvite=yes qualify=yes allow=all [2012] ;grandstream1 type=friend username=2012 secret=1945 nat=1 host=dynamic dtmfmode=rfc2833 canreinvite=yes reinvite=yes qualify=yes allow=all ***************************** and with this extensions.conf file: [general] static=yes writeprotect=yes autofallthrough=yes [globals] CONSOLE=Console/dsp ; Console interface for demo CONSOLE=Zap/1 CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2...

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have other contexts that get included into [local] for long distance and freefone numbers. At a guess would I put the code below in extensions.co...

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

...dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is supposed to be. I am thinking it is a problem between the zap interface and the PSTN. thanks extensions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] ATTENDANT=1001 OUTBOUNDTRUNK=ZAP/g1 [extentions] exten => _10XX,1,Ringing exten => _10XX,2,Dial(SIP/${EXTEN},20) exten => _10XX,3,Answer exten => _10XX,4,VoiceMail(u$...

hello friends, is there any way to simplify that extensions.conf file? [miprimerejemplo] exten => 20000,1,Dial(SIP/20000,30,Ttm) exten => 20000,2,Hangup exten => 20000,102,Voicemail(20000) exten => 20000,103,Hangup exten => 20100,1,Dial(SIP/20100,30,Ttm) exten => 20100,2,Hangup exten => 20100,102,Voicemail(20100) exten => 2010...

...him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my extensions.conf [incoming-calls] exten => _4078698350,1,Goto,bpns-external|${EXTEN}|1 exten => _4078698353,1,Goto,demo1-external|${EXTEN}|1 exten => _4078698359,1,Goto,demo2-external|${EXTEN}|1 exten => _4078698360,1,Goto,demo3-external|${EXTEN}|1 [outgoing-calls] exten => _40...

...'Local/247110358' part, why the hell is that number there? that is the phone number of my neighbor (though without international code), why is it showing there and trying to forwrd it there? below is the whole CLI output i had when i saw this and furhter down the buttom are my sip.conf and extension.conf, i was also trying to set up my voicemail, but somehow that also doesn't work... anyhow nay help is appreciated, thx a bunch katie-jody :D -- Executing NoOp("SIP/9083XXX-0816b208", "Incoming-s. CallerID:"0031648978254" <0031648978254> - Calling:s&qu...

Hi all, i;m obviously a newbie, its been 2 days that im trying to figure out a way to deny a specific extension (300) from calling another specific extensions (03) except if the caller punch a specified password.. sorry if im not explaining myself well.. heres an example: i called my pstn line(with 300 as its sip account), an attendant answers and asks me to punch in an extension number right now if i d...

...=fwd.pulver.com dtmfmode=inband nat=yes canreinvite=no [101] disallow=all allow=ulaw type=friend host=dynamic dtmfmode=inband username=101 secret=testing123 context=home nat=no [102] disallow=all allow=ulaw type=friend host=dynamic dtmf=inband username=102 secret=testing123 context=home nat=no ; Extensions.conf [general] static=yes writeprotect=no [globals] MAINPHONE=SIP/101 FWDUSERID1=533990 FWD1USERNAME=Norman Zhang FWDPREFIX=* HOMENUMBER=XXXXXXXXXX ; Macros [macro-fastbusy] exten => s,1,Answer exten => s,2,Wait,1 exten => s,3,Playback(ss-noservice) exten => s,4,Wait(30) exten...