search for: u2000

...ice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has expired. My hardware is 2 X100P card exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Dial(Zap/1/5551212,10) exten => 2000,3,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup Any ideas?

...-08-14 08:49:46",,"2007-08-14 08:49:56",10,0,"NO ANSWER","DOCUMENTATION" "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-1 ","SIP/600-08e0b990","VoiceMail","u2000","2007-08-14 08:50:37","2007-08-14 08:50:52","2007-08-14 08:51:00",23,8,"ANSWERED","DOCUMENTATION" "","07xxxxxxxxx","2000","incomming_pstn","07xxxxxxxxx","IAX2/ubigradin-2 ","SIP/...

...Extension.conf [general] static=yes writeprotect=yes [outgoing] ignorepat => 9 exten => _9NX.,1,Dial(ZAP/1/${EXTEN:1},60,t) exten => _9NX.,2,Congestion [from-sip] exten => 2000,1,NoOp("call for "${EXTEN}) exten => 2000,2,Dial(SIP/2000,20,tr) exten => 2000,3,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,NoOp("call for "${EXTEN}) exten => 2001,2,Dial(SIP/2000,20,tr) exten => 2001,3,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(...

...with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) -------------- next part --------------...

I am a Rails noob, working on bringing a legacy db system into the 21st century. I want to localize the UI for two languages (EN/JA). Since I am already using the salted user system, I have been playing around with T. Fuchs'' localization module. Nice and simple, already up and working. But here''s the problem: Japanese text is wrapping weirdly in table cells. I could set

...xten => 7,3,VoicemailMain [local] include => voicemail ; SIP Phone Operator Office exten => 1000,1,Dial,SIP/opoffice|30 exten => 1000,2,Voicemail,u1000 exten => 1000,102,Voicemail,b1000 ; SIP Phone Operator Field exten => 2000,1,Dial,SIP/opfield|30 exten => 2000,2,Voicemail,u2000 exten => 2000,102,Voicemail,b2000 approach 2: ======== I moved the opoffice Snom to the office side and moved the configurations for this phone in /etc/sip.conf and /etc/extensions.conf to the Asterisk at the office side. Then I added the following line to /etc/extensions.conf to the local...

...--+---------------+----------------+ | id | context | exten | priority | app | appdata | +----+----------+-------+----------+---------------+----------------+ | 1 | from-sip | 2000 | 1 | Dial | SIP/2000,20 | | 2 | from-sip | 2000 | 2 | Voicemail | u2000 | | 3 | from-sip | 2000 | 102 | Voicemail | b2000 | | 4 | from-sip | 2000 | 103 | Hangup | | | 5 | from-sip | 2001 | 1 | Dial | SIP/2001 | | 6 | from-sip | 2001 | 2 | Voicemail | u2001 | | 7 | f...

...m-sip mailbox=101 EXTENSIONS.CONF [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. [bogon-calls] [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) Thanls Arslan, ---------...

...s with macro's exten => 2000,1,DBget(temp=SIP/2000) exten => 2000,2,Goto(${temp}|1) exten => 2000,102,Goto(2000|3) exten => 2000,3,DBget(dnd=DND/2000) exten => 2000,4,Goto(2000|6) exten => 2000,104,Goto(2000|5) exten => 2000,5,Dial(SIP/2000,20) exten => 2000,6,Voicemail2(u2000) exten => 2000,7,Hangup exten => 2000,106,Voicemail2(b2000) ; busy exten => 2000,107,Hangup Thanks, Brian

...ia lists the following codepoints as whitespace (I'm sure there is a more definitive reference but I could not find one with some quick googling): U0009-U000D (Control characters, containing TAB, CR and LF) U0020 SPACE U0085 NEL U00A0 NBSP U1680 OGHAM SPACE MARK U180E MONGOLIAN VOWEL SEPARATOR U2000-U200A (different sorts of spaces) U2028 LSP U2029 PSP U202F NARROW NBSP U205F MEDIUM MATHEMATICAL SPACE U3000 IDEOGRAPHIC SPACE would it be possible for R to treat these all in the same way? (Or does it already but my R is misconfigured?) Hadley -- http://had.co.nz/

...rid="Fred F"<200> dissallow=all Extensions.conf [default] exten => 1000,1,Dial,Zap/1|20 exten => 1000,2,Voicemail,u1000 exten => 1000,3,Hangup exten => 1000,102,Voicemail,b1000 exten => 1000,103,Hangup exten => 2000,1,Dial,Zap/2|20 exten => 2000,2,Voicemail,u2000 exten => 2000,3,Hangup exten => 2000,102,Voicemail,b2000 exten => 2000,103,Hangup exten => _NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) ; Dial Broadvoice for 30 seconds exten => _NXXNXXXXXX, 2, congestion() ; No answer, nothing exten => _NXXNXXXXXX, 102, busy() ; Busy...

I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better

...l allow=gsm allow=ulaw allow=alaw allow=G723.1 context=from-sip [2000] type=friend username=2000 secret=xxxxxxxx host=dynamic mailbox=2000 extensions.conf [general] static=yes writeprotect=yes [globals] PSTN-1=Zap/1 [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) include => to-pstn [to-pstn] exten => _1NXXNXXXXXX,1,Dial(${PSTN-1}/${EXTEN}) exten => _NXXNXXXXXX,1,Dial({PSTN-1}/1${EXTEN}) Can anyone help me out here? Thanks, Scot...

...) exten => s,2,Voicemail,u1030 exten => s,102,Voicemail,b1030 exten => 8500,1,VoicemailMain [demo2-internal] include => demo2-external include => bpns-internal include => outgoing-calls [demo3-external] exten => s,1,Dial(SIP/2000,20,tr) exten => s,2,Voicemail,u2000 exten => s,102,Voicemail,b2000 exten => 8500,1,VoicemailMain [demo3-internal] include => demo3-external include => bpns-internal include => outgoing-calls It doesn't work. I have a couple asterisk guru friends who swear it should work. Here is what asterisk tells me...

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

...ss [iax1Fedora]type=friendsecret=volgausername=iax1Fedoracontext=fedora-contexthost=dynamictrunk=yes-----------------------------------------------;fedora server extension.conf[fedora-context]exten =>2000,1,Set(TIMEOUT(absolute)=144) exten =>2000,2,dial(sip/2000/20)exten =>2000,3,voicemail(u2000 at default) exten=> _3XXX,1,Answer()exten=> _3XXX,2,Dial(IAX2/iax1Centos:volga at 192.168.0.22/${EXTEN}@centos-context)exten=> _3XXX,3,Hangup()-----------------------------------------------;fedora sip.conf [general]port=5060bindaddr=0.0.0.0allow=allvideosupport=yes ;to support video cal...

Hi People, Ok i've tried everything I can think of but cant get this to work. Can someone please give me an example of their sip.conf settings and also the details of the settings in their grandstream phone to allow: 1. Grandstream phone to register with asterisk when on same lan. 2. Grandstream phone to register with asterisk when phone is behind a nat. Regards, Aaron.

...defaultip=192.168.255.54 context=from-sip mailbox=2000@local dtmfmode=info callerid="Me" <2000> My extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,30) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => _9.,1,Dial(SIP/${EXTEN:1}@iconnect,60,r) exten => 2999,1,VoicemailMain(${CALLERIDNUM}) Can anyone tell me if there is some error I am overlooking. I can't see it being bandwidth or the nat gateway since the bud...

...8 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no extension.conf [general] static = yes writeprotect = yes [bogon-calls] exten => _.,1,Congestion [from-iax] exten => 105,1,Dial(IAX2/IaxCocomall1@202.138.166.6,20) ;exten => 105,2,Voicemail(u2000) ;exten => 105,102,Voicemail(b2000) exten => 105,103,Hangup exten => 106,1,Dial(IAX2/IaxCocomall2@202.138.166.6,20) ;exten => 106,2,Voicemail(u2001) ;exten => 106,102,Voicemail(b2001) exten => 106,103,Hangup exten => 107,1,Dial(IAX2/IaxCocomall3@202.138.166.6,20) ;exten =>...

Hi I am trying to make asterisk load config from database, so far I get the sip, extension working, but voicemail seems still looking for config file, not from the database. the extconfig.conf looks like ... sipusers => mysql,asterisk,sip_buddies sippeers => mysql,asterisk,sip_buddies extensions => mysql,asterisk,extensions_table voicemail => mysql,asterisk,voicemail_users .. the