search for: _1nxxnxxxxxx

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten => _1NXXNXXXXXX,100,NoOp(Calling my cell w/special CID) exten => _1NXXNXXXXXX,n,Set(...

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(T...

...ing the ${CALLERIDNUM} variable in the current context (var ${CONTEXT}). exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) include => inbound include => outgoing [inbound] exten => 9045622082,1,Goto(default,101,1) [outgoing] ; The following gives an Unknown Caller ID ;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _NX...

hi, i write this looking for free conference room, i checl code and don?t see any error but die at priority 7 if room 1001 have users in exten => _1NXXNXXXXXX,1,RouteCall(${EXTEN}) exten => _1NXXNXXXXXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13) exten => _1NXXNXXXXXX,3,Setvar,var=0 exten => _1NXXNXXXXXX,4,MeetMeCount(1001|var) exten => _1NXXNXXXXXX,5,GotoIf($[${var} =0]?7:6) exten => _1NXXNXXXXXX,6,Meetme(1001|M) exten => _1NXXMXXXXXX...

...;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup. ;This is where it breaks. I tried to make it so there can't be more than 2 calls on SIP channels at once. ;Since it counts the phone as a channel, and adds it to the group, I had to use 4. [internalphones] exten = _1NXXNXXXXXX,1,Set(GROUP()=SIPGROUP) exten = _1NXXNXXXXXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} >= 4]?100) ;If the group has 2 or more calls, do not dial. exten = _1NXXNXXXXXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)}) exten = _1NXXNXXXXXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${...

...rxgain=6.0 txgain=0.0 faxdetect=no signalling=fxs_ks context=from-zaptel group=0 channel => 2 signalling=fxs_ks context=from-zaptel group=0 channel => 3 ------------------------------- EXTENSIONS.CONF TRUNK_OPTIONS=rTt ;r here because of the 10-12 second delay exten => _1NXXNXXXXXX,1,Dial(ZAP/2/${EXTEN},120,${TRUNK_OPTIONS}) exten => _1NXXNXXXXXX,n,Dial(ZAP/3/${EXTEN},120,${TRUNK_OPTIONS}) exten => _1NXXNXXXXXX,n,Dial(ZAP/4/${EXTEN},120,${TRUNK_OPTIONS}) exten => _1NXXNXXXXXX,n,Hangup() --------------------------- Note that my extensions.conf...

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that doesn't work. Has anybody tried this before?

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

...en there is silence, so maybe it has something to do with that? I have updated to the latest build of app_amd.c and still have had no luck, although originally things would just hang completely on AMD so it was a step in the right direction. :-) Here's my dialplan: [robocop2] exten => _1NXXNXXXXXX,1,Set(CALLERID(number)=6666666666) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)=Robocop) exten => _1NXXNXXXXXX,n,AMD exten => _1NXXNXXXXXX,n,GotoIf($[${AMDSTATUS}=HUMAN]?human:machine) exten => _1NXXNXXXXXX,n(human),Playback(rick-roll-short) exten => _1NXXNXXXXXX,n,Hangup exten => _...

I am wondering if there is a way to create a SIP/IAX group of outgoing lines like Zap groups. I am currently using the following method, but would like to use features such as ?g2? that would list all the accounts for a SIP or IAX connection. exten => _1NXXNXXXXXX,1,Dial(SIP/account_name:Password@gw5.voicepulse.com/${EXTEN }) exten => _1NXXNXXXXXX,102,Dial(IAX2/account_name:Password@gwiaxt02.voicepulse.com /${EXTEN}) exten => _1NXXNXXXXXX,2,Congestion Thanks. Tim. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (htt...

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|500) exten => _1NXXNXXXXXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN}) Is there a way I can create...

Problem: Flash on X100P does not flash. Phone line has Call Transfer. With this line plugged into a regular phone, it can receive a phone call. Then, depress the hook momentarily, release. Dialtone is now available. Dial a different number. Call is answered. Hook Flash again, now in a three way call. Hang up. The other two parties are still in communication. Now, plug same line into the X100P.

...DR doesn't get populated with the destination phone number and the LastApp field get populated with the recording information. Here's my recording stanza. All the support phones are in a context called record-calls so that it happens all the time. ; Record calls [record-calls] exten => _1NXXNXXXXXX.,1,SetVar(CALLFILENAME=${TIMESTAMP}-${CALLERIDNUM}-${EXTEN:2}) exten => _1NXXNXXXXXX.,2,Monitor(wav,${CALLFILENAME}) exten => _1NXXNXXXXXX.,3,Answer exten => _1NXXNXXXXXX.,4,Background(this-call-may-be) exten => _1NXXNXXXXXX.,5,Background(recorded) exten => _1NXXNXXXXXX.,6,Background...

...-sip] exten => 101,1,Dial(SIP/SK-101,20) exten => 101,2,Voicemail(u101) exten => 101,102,Voicemail(b101) exten => 101,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) [sourcekit-main] include=>sourcekit-sip exten => +19193233010,1,GoTo(sourcekit-sip,101,1) exten => _1NXXNXXXXXX,1,SetCallerID(9193233010) exten => _1NXXNXXXXXX,2,Dial(IAX2/netlogic/${EXTEN}) exten => _1NXXNXXXXXX,3,Dial(IAX2/username:secret@iax.simpletelecom.com/${EXTEN}) exten => _1NXXNXXXXXX,4,Hangup [netlogic] include=>sourcekit-main and, thr debug output from * CLI: Asterisk Ready. *CLI&g...

...-- Hungup 'IAX2/cubix-19' Shouldn't that return CONGESTION instead? I thought that NOANSWER was reserved for calls that reach app_dial's timeout limit? Or am I just missing something simple? Here is the relevant extensions.conf logic that I am using [e164] ; Dundi exten => _1NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN}) ; Dispatch First Trunk exten => _1NXXNXXXXXX,2,Macro(nocdial,${TRUNK}/${EXTEN}) exten => _1NXXNXXXXXX,3,ResetCDR ; On Failure, Dispatch Second Trunk exten => _1NXXNXXXXXX,4,Macro(nocdial,${TRUNK2}/${EXTEN}) exten => _1NXXNXXXXXX,5,ResetCDR ; Third time is a...

...78414198,1,Dial(SIP/8478414198) exten => 8478414198,2,Hangup [long-users] exten => 8478414199,1,Dial(SIP/8478414199) exten => 8478414199,2,Hangup [local] exten => _XXXXXX,1,Dial(SIP/${EXTEN}@VGW) exten => _XXXXXX,2,Congestion include => local-users [long-distance] exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@VGW) exten => _1NXXNXXXXXX,2,Congestion include => local include => long-users

...w=alaw allow=gsm mailboxdetail=yes register => meadmaker:<mypassword>@switch-2.nufone.net #include iax_additional.conf #include iax_custom.conf [NuFone] type=peer host=switch-1.nufone.net secret=<mypassword> and here is what I added to extension.conf: ;added for nufone exten => _1NXXNXXXXXX,1,SetCallerID(8662521540) exten => _1NXXNXXXXXX,2,Dial,IAX2/meadmaker@NuFone/${EXTEN} [inbound] exten => 8662521540,1,Answer() exten => 8662521540,2,Wait(2) exten => 8662521540,3,Dial(200) I would appreciate some help. Trying to get this working today. Have wiped out my machine bec...

...simple as it seems? Broadvoice is circut busy? Can any one think of any other reason I might get this message? Or do I just need to call BroadVoice and complain? I have tried two different proxy's (ip's in /etc/hosts) and get the same error. in extensions.conf: [outgoing] exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@proxy.bos.broadvoice.com,30) ; exten => _1NXXNXXXXXX, 2, congestion() ; No answer, nothing exten => _1NXXNXXXXXX, 102, busy() ; Busy in sip.conf: [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 50...

Hi, i am having a problem with my point to point t1, which is being resolved and is a seperate issue. sangoma support has been a huge help and i am waiting on verizon to increase the signal output of the smartjack. But my issue is that in the meantime my fallover extensions arent working. Well they are on the CPE side but not on the NET side. The NET side still thinks its making calls, they

...ing the ${CALLERIDNUM} variable in the current context (var ${CONTEXT}). exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) include => inbound include => outgoing [inbound] exten => 9045622082,1,Goto(default,101,1) [outgoing] ; The following gives an Unknown Caller ID ;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _NX...