search for: 2002,1

...rks. I follow the instructions from http://www.voip-info.org/wiki/index.php page=Asterisk+standard+extensions but it still doesn't work. [2001] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [2002] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [local] exten => 2001,1,Dial(SIP/2001,10,tr) exten => 2002,1,Dial(SIP/2002,10,tr) [notify] exten => 2001,hint,SIP/2001 exten => 2002,hi...

Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back from xxx.xx.xx.xxx -- SIP/2002-f29b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from xxx.xx.xx...

...ailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=cisco host=dynamic ;host=192.168.22.2 context=from-sip mailbox=101 [2002] ; Duplicate of 2000, except with different auth data type=friend username=2002 secret=cisco ;host=192.168.22.3 host=dynamic context=from-sip mailbox=102 extensions.conf [2000] exten => 2000,1,Dial(SIP/2000,20) [2001] exten => 2001,1,Dial(SIP/2001,20) [2002] exten => 2002,...

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I

On machine A I have something like the following in extensions.conf: [iax-extensions] exten => _9.,1,Dial(IAX2/machineB/${EXTEN:1}@mycontext) exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _9.,3,Hangup On machineB I have something like this: [mycontext] exten => 2002,1,Dial(SIP/2002,60) exten => 2002,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => 2002,3,Hangup If I use a phone on machineA to dial 92002, then of course 2002 rings. If 2002 is busy, then on machineB's console I see the NoOp command with DIALSTATUS=BUSY, as expected. However, on machineA...

...2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside secret=1234 host=dynamic [2003] type=friend contex=outside secret=1234 host=dynamic ? [2004] type=friend contex=outside secret=1234 host=dynamic [2005] type=friend contex=outside secret=1234 host=dynamic ? [2006] type=friend contex=internal secret=1234 host=dynamic [...

Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call!

...on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2002] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes ;******************************* ; BEGIN: CUCM(s) Added Below ;******************************* [cucm] host=172.16.200.232 type=friend ;nat=no ;type=peer context=incoming canreinvite=yes ;insecure...

...is card did not connect with E1 line how to loop E400P card to test ? now I loop the card. span 1 ---span2 RJ45 pins 1--4 2--5 but show : When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' Asterisk Ready. *CLI> -- Registered SIP '2002' at 192.168.139.59 port 3289 expires 120 -- Executing Dial("SIP/2002-a4af", "Zap/1") in new stack Jul 28 15:51:25 NOTICE[1910]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Hangup(&...

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

...l support? I'm sure someone who knows what they are doing could have this figured out in an hour instead of me banging my head against the wall for another week. At least that way, I'd have a working system to reverse engineer. Thanks for any help. Steve D. Typical entries: Sip.conf [2002] type=friend username=2002 secret=2002 host=dynamic context=internal mailbox=2002 defaultip=192.168.140.51 callerid="Kevin" <2002> canreinvite=no reinvite=no dtmfmode=inband disallow=all allow=ulaw allow=gsm Iax.conf entry: [2050] type=friend ;accountcode=iaxy host=dynamic username...

...Basis: DAILY OVERLAY(TS21560100, TS21555100) ?Vollzeitaequivalente Beschaeftigung? stands for ?Full-time equivalent employment? Also, I have the same error message when I run the code from the manual: > seriesA <- tis(1:24,start=c(2002,1),freq =12) > seriesB <- tis(1:104, start = c(2002, 1), tif = "wmonday") > documentation(seriesB) <- paste("Line", 1:4, "of seriesB documentation") > putfame(c(mser = "seriesA", wser = "seriesB"), db = "myfame.db") Warn...

...2.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24 srvlookup = yes allow=all [2001] ;grandstream 2 type=friend username=2001 secret=1945 canreinvite=yes reinvite=yes host=dynamic dtmfmode=rfc2833 qualify=yes ;mailbox=2001 nat=1 allow=all [2002] ; soft phone type=friend username=2002 secret=1945 canreinvite=yes reinvite=yes host=dynamic dtmfmode=rfc2833 qualify=200 mailbox=2002 nat=1 allow=all [2010]; wi-fi phone type=friend username=2010 secret=1945 nat=1 host=dynamic dtmfmode=rfc2833 canreinvite=yes reinvite=yes qu...

...ndaddr = 0.0.0.0 context = bogon-calls disallow = all nat=yes canreinvite=yes allowguest=no allow=ulaw allow=alaw allow=g711 allow=g729 allow=gsm allow=ilbc [2000] type=friend context=my-phones secret= allow=ulaw host=dynamic [2001] type=friend context=my-phones secret= allow=ulaw host=dynamic [2002] type=friend context=my-phones secret= allow=ulaw host=dynamic [2003] type=friend context=my-phones secret= allow=ulaw host=dynamic [xxx.xxx.xxx.yyy] context=pstn-incoming type=friend host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very [1001]...

I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some???

...2020,4,Queue(MyQueue) exten=> 2020,5,Hangup my agent 1001 and 1002 has been login (in file extensions.conf) [ext-agents] ;Agent Login ;exten=> 2001,1,AgentCallbackLogin(|${CALLERIDNUM}@ext-agents) exten=> 2001,1,AgentCallbackLogin(|${CALLERIDNUM}@from-internal) ;Agent Logout exten=> 2002,1,AgentCallbackLogin(|l) my dial option as DIAL_OPTIONS = twWrm in features.conf [featuremap] automon=1 when from telco, i dial to asterisk, then press 2020, agent ringing and agent 1001 accept the call, from agent 1001 i cannot packing call or transfer call or monitor call. if i don't u...

...gittimeout = 60 digittimeout = 8 autoanswer_ring_time = 0 autoanswer_tone = 0x32 remotehangup_tone = 0x32 transfer_tone = 0 callwaiting_tone = 0x2d musicclass=default language=en rtptos = 184 echocancel = on silencesuppression = off trustphoneip = no tos = 0x68 [devices] type = 7960 autologin = 2002 description = phone2002 dtmfmode = inband imageversiom = P00307020200 dnd = on trustphoneip = no speeddial = 2000 private = on device => SEP00036BC3852B [lines] id = 2002 ; future use pin = 1234 ; future use label = 2002...

Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? [from-sip] exten => 2999,1,VoicemailMain(${CALLERIDNUM}@vulpes) exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(2000@vulpes)...

...rough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten => 1004,1,Dial(SIP/1004,,t) exten => 2001,1,agi,login.php exten => 2002,1,Queue(FAO|tT) exten => 2004,1,MusicOnHold exten => 2004,2,Hangup When I call from 100001000 to 1001, I can press # and type 2004 to transfer and 100001000 gets MOH. When I dial 2002 (queue) from 100001000, 1001 rings and I am able to talk both ways, but nothing happens when I press # at 10...

...tions] 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 => 2002,1,Dial(IAX2/iaxphone,20) ;exten => 2002,2,Voicemail(u2002) ;exten => 2002,102,Voicemail(b2002) ;exten => 2002,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) exten => h,1,Hangup exten => t,1,Hangup exten => i,1,Playback(invalid) exten => i,2,Goto(default,main,3) e...