search for: 1001,2

...command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan should jump to the next priority. exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})}) exten => 1001,2,,NoOP(${DIALSTATUS}) exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1) exten => h,1,NoOp() exten => h,n,NoOP(${DIALSTATUS}) ----------------------------------------------------------------------- If i try to dial the same...

.... Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/guest@24.202.159.205/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten =>...

...XTEN},60) ;voipjet NANPA exten => _011.,1,Dial(IAX2/402@voipjet/${EXTEN},60) ;voipjet WORLD [bpns-external] exten => s,1,Playback,bpnsmenu exten => 1,1,Dial(SIP/1003,20,tr) exten => 1,2,Voicemail,u1003 exten => 1,102,Voicemail,b1003 exten => 2,1,Dial(SIP/1001,20,tr) exten => 2,2,Voicemail,u1001 exten => 2,102,Voicemail,b1001 exten => 3,1,Dial(SIP/1002,20,tr) exten => 3,2,VOicemail,u1002 exten => 3,102,Voicemail,b1002 exten => 1001,1,Dial(SIP/1001,20,tr) exten => 1001,2,Voicemail,u1001 exten => 1001,102,VOicemail,b1002 ex...

...l to an available local rep in another state. I thought this was possible .... until I realized the "transfer" only works on xPRO, which isn't available for linux. So I cant rely on SIP to handle this, I set up my extensions.conf have transfers, ie: [sip-exten] exten => 1001,1,Dial(SIP/1001,20,Trt) exten => 1001,2,Hangup And features.conf is : [featuremap] blindxfer => *1 ; Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record atxfer => *2 ; Attended transfer OK...

...=> _1866NXXXXXX,1,Dial(IAX2/jmproductions:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1800NXXXXXX,1,Dial(IAX2/jmproductions:xxxxx@iaxtel.com/${EXTEN}@iaxtel) [sip] include => iaxtel exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => s,1,Dial(SIP/one|20|tr) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/guest@24.202.159.205/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten =>...

...bx.c, Line 1171 (pbx_extension_helper): Cannot find extension context 'from-sip' DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '746374551@10.0.1.5' of Response 4964: Not Found This is my extensions.conf file: [general] [from-sip] exten => 1001,1,Dial(sip/1001@10.0.1.5,20) exten => 1001,2,Voicemail(u1001) exten => 1001,102,Voicemail(b1001) exten => 1001,103,Hangup exten => 1002,1,Dial(1002,20) exten => 1002,2,Voicemail(u1002) exten => 1002,102,Voicemail(b1002) exten => 1002,103,Hangup And this is my s...

Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary

...lobals] [sip] exten => 301,1,Dial(SIP/Nick,20,tr) exten => 302,1,Dial(SIP/Sharon,20,tr) exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr) exten => 302,2,VoiceMail,u302 exten => 301,2,VoiceMail,u301 exten => 1000,2,VoiceMail,u9999 exten => 1000,102,VoiceMail,b9999 exten => 1001,1,Ringing exten => 1001,2,Wait(2) exten => 1001,3,VoicemailMain include => outgoing [incoming] exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr) [outgoing] exten => _7.,1,Dial(IAX2/login:passwd@XXX.XXX.XXX>XXX/${EXTEN:1}) exten => 5.,1,Dial,Zap/1/${EXTEN:1} [9103] exten =>...

...fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=2 txgain=2 group=1 channel => 1-7 extensions.conf ... [from-sip] ignorepat => 9 exten => _9NXXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _91XXXNXXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) ; generic phone extension exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,VoiceMail(u1001) exten => 1001,102,VoiceMail(b1001) exten => 1001,103,Hangu ... sip.conf ... [1001] type=friend username=1001 fromuser=1001 callerid=User Name <1001> host=dynamic nat=yes canreinvite=yes dtmfmode=info mailbox=1001@default disall...

....conf: [general] static=yes writeprotect=no [globals] [sip] exten => 1,1,Dial(SIP/phone1,20,tr) exten => 2,1,Dial(SIP/phone2,20,tr) exten => 2,2,VoiceMail,u1234 exten => 2,102,VoiceMail,b1234 ;exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) exten => 1001,1,Ringing exten => 1001,2,Wait(2) exten => 1001,3,VoicemailMain,s1234 exten => 6601,1,WaitMusicOnHold(60) exten => 232999,1,Dial(SIP/phone1,30,tr) exten => 232999,2,Hangup I am behind a NATed fire wall, but I'm not sure that is related. Any ideas or help (working...

...ls to work. So I have gone back to a very basic FWD config, with one phone which as far as I am aware should work, but doesn't. I cannot find info on how to fix this. Below is my sip.conf [general] port = 5060 bindaddr = xxx.xxx.xxx.xxx context = sip register => 2xxxx:xxxx@fwd.pulver.com/1001 [fwd] type=friend secret=xxxxxx username=xxxxxx host=fwd.pulver.com ; ; [1001] type=friend username=xxxxxx host=dynamic secret=xxxxxxx callerid=Home <1001> dtmfmode=RFC2833 mailbox=1001 context=sip and here is my extensions.conf: [general] static=yes writeprotect=no ; [globals] HOME=S...

...] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm) exten => *0,3,Playback(fcopba1) exten => *0,4,Hangup() include => extentions include => pasvalide [extentions] exten => 1001,1,Dial(SIP/sipchan1001,10) exten => 1001,2,Voicemail(u1001) exten => 1001,3,Hangup() exten => 1002,1,Dial(SIP/sipchan1002,10) exten => 1002,2,Voicemail(u1002) exten => 1002,3,Hangup() exten => *,1,VoicemailMain(${CALLERIDNUM}) ;exten => *,1,VoicemailMain() exten => *,2,Hangu...

...= 7910 autologin = test1 description = Test2 7910 context = sccp [SEP0005323DB87B] type = 7910 autologin = test2 description = Test2 7910 context = sccp [SEP0002B9A754BD] type = 7960 autologin = test3 description = Test3 7960 context = sccp [test1] id = 1001 label = Test1 description = Test1 context = sccp callwaiting = 1 mailbox = 1001 callerid = "Test Line 1", <1001> [test2] id = 1002 label = Test2 description = Test2 context = sccp callwaiting = 1 mailbox = 1002 callerid = "Test Line 2...

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I haven't been able to find anything indicating whether or not I need to configure it somewhere or if there are any special FreeBSD kernel...

...oks in the > directory called "default", or is there a way to make MWI look in another VM > directory. > > thanks > > lee goodman > > > voicemail.conf > [general] > format=wav > maxmessage=180 > [sip] > 1000 => 1000,LG,xxxx@comcast.net > 1001 => 1001,TG,yyyyy@comcast.net > 1002 => 1002,BG,zzzzz@comcast.net > > extensions.conf > > [incoming] > exten => s,1,Background(goodmanmenu) > exten => s,2,DigitTimeout,5 > exten => s,3,responsetimeout,10 > exten => 1000,1,Goto(sip,1000,1) > exten =&g...

...ones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting) exten => s,6,BackGround(auto-menu) [ciscophones] exten => 1000,1,Dial(SIP/1000,15) exten => 1000,2,Voicemail(u1000) exten => 1001,1,Dial(SIP/1001,15) exten => 1001,2,Voicemail(u1001) exten => o,1,Goto(incoming,s,6) exten => 0,1,Goto(ciscophones,1001,1) Now if I press "0" during the voicemail prompt - it will Dial extension 1001 instead of routing to the [incoming] context and re-play the auto-menu. Does...

...placing a call on the cmd line saying: > NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to > extension '91AreaCodePhone#' rejected because extension not found. > > > What I have in Extensions.conf is: > [gary-incomming] > exten => 1001,1,Dial(IAX2/gogh) > exten => 1001,2,HangUp() > exten => 120,1,Dial(SIP/Gary) > exten => Gary,1,Goto(120,1) > exten => i,1,Playback(invalid) > exten => i,2,Goto(s,1) > exten => s,1,Wait(1) > exten => s,2,Answer() > exten => s,3,NoOp(${CALLERID}) > e...

...7 exten => 7777,3,Hangup exten => 7777,102,VoiceMail,b7777 exten => 7777,3,Hangup exten => 7777,103,Hangup exten => 8888,1,Playback(transfer,skip) exten => 8888,2,Ringing exten => 8888,3,Wait(2) exten => 8888,4,VoiceMail,u8888 exten => 8888,104,VoiceMail,b8888 exten => 1001,1,Ringing exten => 1001,2,Wait(2) exten => 1001,3,VoicemailMain exten => _NXXNXXXXXX,1,Background(beep) ;exten => _NXXNXXXXXX,2,SayDigits(${EXTEN}) ;exten => _NXXNXXXXXX,3,Goto(testdtmf|s|1) exten => t,1,Ringing exten => t,2,Hangup exten => i,1,Ringing exten => i,2,Hangup...

...ello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to call F, and it will ring. Voice from softphone to F carries over and I can hear it; however, no voice from F to softphone will carry. I have been experimenting with different codec and other cisco/asterisk...

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go down. And Icould not leave a messages. Please teach me how to resolve this problem. Here is c...