similar to: MEMBERINTERFACE and MEMBERNAME questions

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! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

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 =>

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

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

Hi! Have a weired problem with Avaya 9650 phones: extensions.conf exten => 0317998975,hint,SIP/0317998975 exten => 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1) exten => 0317998975,2,Hangup() exten => 0317998975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 0317998975-INUSE,2,Hangup() exten => 0317998975-NOANSWER,1,VoiceMail(0317998975 at

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 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! 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:

Hi, I have missed something so I wonder if someone could explain for me? 0424449647 desktop phone 0106024647 DECT phone 0424449630 Helsingborg queue extensions.conf --------------- [support] exten => 0424449647,hint,SIP/0424449647&SIP/0106024647 exten => 0424449647,1,Dial(SIP/0424449647&SIP/0106024647,15,rtT) [inputinterior.se] exten => 0/0424449647,1,Answer() exten =>

Hi all! First of all i would like to say that i am not a Samba4 guru so my question may be "stupid". I have 2 Samba4 servers up and runnning: Server 1: netbios name = PDC workgroup = GBG realm = GBG.INPUTINTERIOR.SE server role = domain controller Server 2: netbios name = PDC workgroup = MLM realm = MLM.INPUTINTERIOR.SE server role = domain controller Here comes my

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

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

Hallo, ABCTI (an open-source CTI client for Asterisk) has moved to beta stage. Find it on: http://abcti.sourceforge.net For the first time, we now have windows installers that actually work ;-) We would appreciate any feedback you can give. Regards, -- o

Hi! I try to get rid of some part of CALLERID(name) but I cant realy figure out a way to do it. For example: CALLERID(name) = "Martela (fax)" I am just looking for the part before ? (? in my case ?Martela?. I can?t serch for ? ?, could be many ? ?, but only one ? (?, thought i could do something like: exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1}) But that gave me

Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

I have running an Asterisk 1.4.20 box and I want to get the value of the MEMBERINTERFACE variable to know about which queue member received the call. I set 'setinterfacevar=' to 'yes' in queues.conf but I can?t get this value.

Asterisk 12.5 I'm using AMI to initiate a "call me now" feature from the web site. The AMI looks like: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same