search for: _2xx

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten => _2XX,4,SIPAddHeader(Call-Info: <sip:83.222.226.126>\;answer-after=0) exten => _2XX,5,Page(SIP/winsor_${EXTEN}) I found a...

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank you, Alex Fedorov

...p() [spa] exten =>_301,1,GoTo(sipura-line,${EXTEN},1) exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten => _1XX,2,VoiceMail(${EXTEN}@default) ; direct 2 voicemail box if line is busy or unavailable exten => _1XX,3,HangUp() exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times exten => _2XX,2,VoiceMail(${EXTEN}@default) ; directs to voicemail box if line is busy or unavailable exten => _2XX,3,HangUp() exten => _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals to 5 seconds so it will rin...

...MESTAMP}-${UNIQUEID}.wav ) ; Removed exten 201 and replaced with 238 on 8/24/06 ;exten => s,n,Dial(Sip/201&Sip/209&Sip/211|20) exten => s,n,Dial(Sip/238&Sip/209&Sip/211|20) exten => s,n,Dial(Sip/227&Sip/225&Sip/213|20) exten => s,n,Goto(ivr,s,1) exten => _2XX,1,Macro(extensions,${EXTEN}) exten => 1234,1,Dial(Sip/phone1,20) ;Aastra 480iCT [closed] exten => s,1,Goto(ivr,s,1) [ivr] exten => s,1,Answer exten => s,n,Set(LOOPCOUNT=0) exten => s,n(begin),Set(TIMEOUT(digit)=3) exten => s,n,Set(TIMEOUT(response)=10) exten =&gt...

I have two calls (626 and 542) coming into the same phone (524). SIP/524-000005b5!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!9!SIP/542-000005b4 SIP/542-000005b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-000005b5 SIP/524-000005b3!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!40!SIP/526-000005b2 SIP/526-000005b2!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!209!!3!40!SIP/524-000005b3 One is on hold and one...

...; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/G2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [default] ; DGB [internal] exten => _2xx,1,Dial(SIP/${EXTEN},15,tTm) exten => _2xx,2,VoiceMail(${EXTEN}@voicemail) exten => _2xx,3,Playback(vm-goodbye) exten => _2xx,4,Hangup exten => *98,1,Answer exten => *98,2,Wait(1) exten => *98,3,VoiceMailMain(${CALLERID}@voicemail) exten => *98,4,Hangup exten => *600,1,Answ...

Hi, I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice 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

Hey everyone, I have a more of an opinion question then a technical question. The asterisk server I am setting up is going to host 3 different businesses. Each business is in the same building, and on the same network. My question is regarding calls coming in and going out. We are a small ISP and have a lot of numbers that are forwarded to our phone system. The other companies have about 3

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

...application like this is my extensions.conf: > > exten => _0X.,1,Answer() > exten => _0X.,n,MixMonitor(${CALLERID(num)}-${STRFTIME($ > {EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav) > exten => _0X.,n,Dial(IAX2/pabx-canall/${EXTEN},60,tT) > > exten => _2XX,1,Answer() exten => _2XX,n,MixMonitor(${CALLERID(num)}-$ > {STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav) > exten => _2XX,n,Dial(SIP/${EXTEN},60,tT) > > The scenario is as following: > > 1) 201 asks operator for an external call, hangs up. The au...

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

...22 secret=xxxxxxxxx host=dynamic call-limit=3 qualify=yes nat=yes dtmfmode=rfc2833 mailbox=422 vmexten=702 fromdomain=sip3.xxxxx.co.uk [223] type=friend username=223 secret=xxxxx host=dynamic call-limit=3 qualify=yes nat=yes dtmfmode=rfc2833 extensions.conf [default] include => blf exten => _2XX,1,SIPAddHeader("Alert-Info:<http://nohost>\;info=alert-internal\;x-line-id=0") exten => _2XX,n,DIAL(SIP/${EXTEN},,tT) exten => _2XX,n,Hangup [blf] exten => 220,hint,SIP/220 exten => 221,hint,SIP/221 exten => 222,hint,SIP/222 exten => 223,hint,SIP/223

...ssword@overhere.overthere/${EXTEN:1}) exten => _9X.,2,Congestion exten => _9X.,3,Hangup [atp-in] exten => 30182849,1,SetMusicOnHold(record) exten => 30182849,2,Dial(SIP/bt-rlm,45,t) exten => 30182849,3,Voicemail,u550 exten => 30182849,103,Voicemail,b550 [te405p-in] exten => _2XX,1,Dial(Zap/g4/${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 inclu...

...2. I then created IAX trunks for each box using 999 as username and password, hostname/IP was set to be other box's IP 3. when done, from the system status panel, I saw the trunks successfully registered to the other box 4. then I added "Outgoing Call Rules" to each box: for box1, _2XX --> to_box2_trunk for box2, _1XX --> to_box1_trunk This setup works ok, the only problem is caller id, i.e. when extension(200) from box2 calls to extension(100) from box1, the call can be made but the caller id displayed on 100 is 999 not 200. I have been on this problem for some time...

Here's an interesting issue we're facing... We would like users to be able to use softphones from home/work and to use their same extensions they do at work. The first step of getting the phones to log in as their same extensions as work is easy and works. However, on the database side, once the client closes, the sip table is cleared of the ip to the phone. This means that no calls are

...ATUS},1) exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u) exten => s-NOANSWER,n,Hangup exten => s-BUSY,1,Voicemail(${MACRO_EXTEN}@voicemail,b) exten => s-BUSY,n,Hangup exten => s-CHANUNAVAIL,1,Playback(pbx-invalid) [from-internal] ; Llamadas a extensiones SIP exten => _2xx,1,Macro(dial,SIP/${EXTEN}) exten => _2xx,n,Hangup exten => 300,1,Dial(SIP/300,30,tTrm) ; Extension analogica exten => 402,1,Macro(dial,DAHDI/2) exten => 402,n,Hangup ; Directorio de extensiones exten => *400,1,Directory(voicemail,from-internal) ; Musica en espera exten => *300...

Hi, I need to connect 2 asterisks in 2 different countries (A and B) for one company so it's possible to make connections between the 2 offices. For connectivity reasons (NAT traversal) i want to connect the 2 asterisk with IAX so that when a user on office A connects via SIP to user on office B the call is going trought IAX channel. Can anyone give me an ideia how to accomplish

I am still testing Asterisk, but I am running in to a lot of problems. I set up numerous extensions, but Asterisk is not performing to tasks correctly. Here is an example. exten => 231,1,Dial(Zap/g1/231|3) exten => 231,2,Voicemail(u231) exten => 231,3,Hangup When I call in and enter extension 231, my call is routed to the correct extension, but it just keeps ringing. If I change the

...Extensions.conf [internal] exten => i,1,Playback(invalid) exten => i,2,Hangup exten => t,1,Hangup exten => 099,1,Echo ;simple echo test when you dial 099 on your phone [outgoing] exten => _1XX,1,Dial(H323/${EXTEN}@192.168.254.250) ; 1xx extension to Salisbury exten => _2XX,1,Dial(H323/${EXTEN}@192.168.20.250) ; 2xx extension to Marcoola exten => 610,1,Dial(H323/${EXTEN}@192.168.30.250) ; 610 to Jindalee exten => 620,1,Dial(H323/${EXTEN}@192.168.40.250) ; 620 to Batteryhill exten => _54XXXXXX,1,Dial(H323/${EXTEN}@192.168.20.250) ; 54 to Marcoola exten...

...be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Then in extensions.conf I have: [default] ; this below for internal extensions - works OK exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm) ; for dialing outbound - over ISDN line - this bit does not work exten => _9XX.,2,Dial(ZAP/g1/${EXTEN},60) exten => _9XX.,2,Hangup Error I get is: -- Executing Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stack Jul 25 11:56:33 NO...