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...! 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-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten => 975-NOANSWER,1,VoiceMail(0317998975 at inputinterior.se,us) exten => 975-NOANSWER,2,Hangup() exten => 975-NOT_INUSE,1,Dial(SIP/0317998975&H323/00733025975 at Avaya,20) exten => 975-NOT_INUSE,2,Goto(975-${DIALSTATUS},1) exten => 975...

...,rtT)) exten => 0320209030,n,Queue(Kinna,nrtT) exten => 0320209030,n,NoOp(${MEMBERINTERFACE}) exten => 0320209030,n,NoOp(${MEMBERNAME}) exten => 0320209030,n,Queue(Goteborg,rtT) exten => 0320209030,n,Hangup() CLI> ---- == Using SIP RTP CoS mark 5 -- Executing [0320209030 at inputinterior.se:1] Answer("SIP/0317998985-00000033", "") in new stack -- Executing [0320209030 at inputinterior.se:2] ExecIf("SIP/0317998985-00000033", "0?Queue(Goteborg,rtT)") in new stack -- Executing [0320209030 at inputinterior.se:3] Queue("SIP/0317998985...

...he display on my phone change to "Connected Line 0317998955"? exten => 956,1,Goto(0317998956,1) exten => 0317998956,1,Set(CONNECTEDLINE(all)=Connected Line ) exten => 0317998956,n,Answer() exten => 0317998956,n,Wait(2) exten => 0317998956,n,Hangup() -- Executing [956 at inputinterior.se:1] Goto("SIP/0317998975-00000004", "0317998956,1") in new stack -- Goto (inputinterior.se,0317998956,1) -- Executing [0317998956 at inputinterior.se:1] Set("SIP/0317998975-00000004", "CONNECTEDLINE(all)=Connected Line ") in new stack -- Executing [03179...

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

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 => 985-NOANSWER,2,PlayBack(vm-goodbye) exten => 985-NOANSWER,3,HangUp() 0317998985 is a direct connected SIP phone 0702221448 is a ce...

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 => 0317998975,1,NoOp(0317998975 at inputinterior.se has state ${DEVICE_STATE(SIP/0317998975)}) exten => 0317998975,2,Dial(SIP/0317998975) It does...

...y "fax applications" and when I try to receive a fax: exten => 960,1,Answer() exten => 960,2,Wait(3) exten => 960,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) [Nov 30 09:29:18] WARNING[19893]: pbx.c:3677 pbx_extension_helper: No application 'ReceiveFAX' for extension (inputinterior.se, 960, 3) Can any "guru" guide me what I am doing wrong? Best regards, MAGNUS BENNGRD Direktnr 031-799 89 75 F?ltspatsgatan 2 421 30 V?stra Fr?lunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: ------ [1] http://www.inputinterior.se -------------- next pa...

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

...X-Lite can dial MeetMe (955) no problem but when 975 dials X-Lite, I get connectio hear X-Lite ringing but Asterisk dumps: -- Registered SIP '0317998985' at 10.242.10.209 port 22796 > Saved useragent "X-Lite release 1103k stamp 53621" for peer 0317998985 -- Executing [985 at inputinterior.se:1] Dial("OOH323/avaya-1", "SIP/0317998985") in new stack == Using UDPTL CoS mark 5 -- Called 0317998985 -- SIP/0317998985-00000001 is ringing Segmentation fault cat /var/log/messages Dec 1 12:02:25 sip2 kernel: [13455.390240] asterisk[15013]: segfault at 0 ip b7edde94 sp...

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 question, can I in some way modify Server 2, so it fetches the user accounts from Server 1? I do want to keep different workgroups. F...

...; 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 /var/spool/asterisk/tmp/fax.pdf magnus.b at inputinterior.se < /dev/null) I do receive the fax, the fax got converted to a pdf but 101,5 never get "executed", when i look in cli, last line is 101,4... can any1 se why? 2:nd example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.t...

...Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscri...

...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 answer_call: Dropping call because extensions '977', 's' and 'i' doesn't exists in context [inputinterior.se] Rewriting of outgoing is working... snip exten => _0X!/0317998975,1,Set(CALLERID(number)=317998975) exten => _0X!/0317998977,1,Set(CALLERID(number)=317998977) exten => _0X!/0317998978,1,Set(CALLERID(number)=317998978) exten => _0X!/0317998985,1,Set(CALLERID(number)=317998985) exte...

Hi, I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. Below example is for a ?Swedish? cell phone, dont know if it works in the same way for other countries. I could define ?redirecting? numbers for 3 traffic cases when u dial my mobile (073-302 59 75): NOT_INUSE call forward to A INUSE

I am using H.323 to create a trunk between Asterisk and Avaya IP Office system. Everything is working correctly, Asterisk can call Avaya and vise versa. Now I create a conference room with a user pin in Asterisk. Avaya can call into the conference room, but can enter the pin number. The error message they are getting is the "invalid pin number". However the pin number works if they

...egy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975&SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: "exten => kalle,hint,SIP/0317998975&SIP/0317998985" just for testing purposes, that did work: kalle at inputinterior.se : SIP/0317998975&SIP/0 State:Idle Watchers 0 But I cant figure out howto "connect" the queue with "kalle", or maybe it is not possible? /Magnus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-u...

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

...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 => 0/0424449647,n,RemoveQueueMember(Helsinborg,Local/0424449647 at support) exten => 0/0424449647,n,Hangup() ; exten => 1/0424449647,1,Answer() exten => 1/0424449647,n,RemoveQueueMember(Helsinborg,Local/0424449647 at support) exten => 1/0...

I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back.