similar to: Manager command that equal to database show CFIM

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi! Trying to figure out how to rewrite calling number of an incoming call... A cell phone (0733025975) dials a X-Lite (977). X-Lite "shows" 733025975 at the display, but I want it to be 0317998975. I thought i could do something like: exten => 977/733025975,1,Set(CALLERID(number)=0317998975) exten => 977,n,Dial(SIP/0317998977) [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272

Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI> queue show 0317998989 0317998989 has 0 calls (max unlimited) in

I set up a call forwarding script in extensions.conf which uses the AstDB but I'm wondering if I reboot the server, will the entry in AstDB still reside? What the script does is when a call comes in, it check to see if there is a null value or a call forward number. If null, it will call the local office connections. If there is a number, it calls that. Now I just need to know if I reboot

Dear All, I'n trying to make a simple call forwarding, however I have small problem when evaluating an expresion. Here is my extensions.conf ... ; Unconditional Call Forward exten => _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten => _#21*X.,2,Hangup() exten => #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})}) exten => #21#,2,Hangup() ... exten =>

Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten =>

Hi! Trying to understand how wrapuptime is working... I have written a small php script that let agents log in/out off a queue. That part is working as a clock but wrapuptime is not doing what I expect. Input Interi?r - Queue Manager 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (5s holdtime, 94s talktime), W:0, C:8, A:1, SL:0.0% within 0s Members: SIP/0317998971

Hi, I have following one-line macro extension: ------------------------ [macro-oneline] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Device(s) to ring ; #exten => s,1,AGI(misterhouse.agi,"CallerID") exten => s,1,NoOp exten => s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, goto 103 exten => s,3,Dial(Local/${temp}@default/n) ;

Hi, I tried to perform call forward in asterisk by writing the following in the dial plan.The data base is getting updated with the caller ID number how ever the call is not getting forwarded. [apps] exten => _*21*XX,1,Set(DB(CFIM/${CALLERID(number)})=${EXTEN:4}) exten => _*21*XX,2,Hangup exten => #21#,1,DBDel(CFIM/${CALLERID(num)})=${EXTEN:4} exten => #21#,2,Hangup Regards, Aparna

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

I have the standard script for activating call forward and when I do a database show, I indeed see: /CFIM/2000 :12125553434 so I presume that means call forwarding is in effect. However, when anyone dials extension 2000, it rings and no forwarding takes place. Is there something basic I'm missing here? Does one have to define, first, what CFIM is? The Newbie Thanks! B.

Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But "core show applications" doesnt show me any "fax applications" and when I try to receive a fax:

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hello, There is a problem that i can not figure out how to solve. I got users with 5 digit usernames for sip. Some users has a callerid for outside calls. I have such problems When a user activates (for ex) call forwarding, System creates that entry on database as CFIM/callerid not the username, So this rule works only if a call is made from outside to the callerid. Not the local calls made

Took some examples from voip-info.org to deal with call forwarding etc: exten => _*21*X.,1,NoOp(Unconditional Call Forward on extension ${CALLERID(num)} to ${EXTEN:4}) exten => _*21*X.,n,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten => _*21*X.,n,Hangup() Problem is that * don't hangup cleanly: Spawn extension (default, *21*2403, 3) exited non-zero on 'SIP/2401-081e7048'