search for: 1001,102

.... 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...

...=> _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...

...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...

...= 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...

...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...

I'd like to know how everyone else is going about getting correct CDR information for calls. Right now I notice that if a call come in and gets parked the CDR info doesn't how the correct info on who picked that call up, also when someone transfer a call there isn't a new record being made so the duration of the call shows up for who received the call and transferred it. I started