search for: 102,30

Displaying 15 results from an estimated 15 matches for "102,30".

2004 Nov 27
2
capi question
...if the (callgroup=0) and fallback to extension 's' issue is a problem, or just asterisk being very verbose... -- started pbx on channel (callgroup=0)! == Starting CAPI[contr1/368466]/33 at isdn,368466,1 failed so falling back to exten 's' -- Called 101 -- Called 102 -- SIP/101-1b74 is ringing -- SIP/102-b2b1 is ringing (extensions.conf) [isdn] exten=> s,1,Dial(SIP/101&SIP/102,30,tr) exten=> s,2,capiCD(020712341234) --also am i correct in thinking that this capiCD line will bounce the 'third' call on my 2 channel line to the numb...
2007 May 10
1
AT530 Telephone
Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten => 105,1,Answer exten => 105,2,Background(/home/user/suport) exten => 1,1,Dial(SIP/101,30,Ttm) exten => 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option "1" it doesn't redirect to 101 extension. Otherwise with the X-Lite extension I select "1" or "2" options and it works perfectly. Anyone has the same problem? I must push another...
2007 Jun 12
1
call from ISDN
...lled g1/943833473 -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup("SIP/101-f9eb", "") in new stack == Spawn extension (SOME, 943833473, 102) exited non-zero on 'SIP/101-f9eb' My extensions.conf is this one: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup exten => 103,1,Dial(SIP/103,30,Ttm) exten =...
2007 Jun 19
1
problem with mISDN
Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten =>_9XXXXXXXX,1,Dial(mISDN/1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() [default] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Dial(SIP/101,30,...
2007 Apr 27
1
can´t anserd the call
...', but no invalid handler -- Hungup 'Zap/1-1' mi configuration files are this: extensions.conf: [general] static=yes writeprotect=yes ;autofallthrough=yes ;clearglobalvars=no ;priorityjumping=no [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup [incoming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,Dial(SIP/101,30,Ttm) [outgoing] exten =>_94XXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW) exten =>_94XXXXXXX,2,Hangup() exten =>_94XXXXXXX,102,Hangup() zapata.conf: [ch...
2007 Jun 13
2
mISDN problem
...- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] --> we have already send Release_complete == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup("SIP/101-081805b8", "") in new stack == Spawn extension (SOME, 943833473, 102) exited non-zero on 'SIP/101-081805b8' I dont't know what happen. Some can help me??? Thanks to everybody. How can I saw the status of the ISDN??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachme...
2006 Nov 19
2
WaitExten only reading 1 digit.
...xten wait for 3 digits? I have setup the extension "100" for users to reach the switchboard as they would from outside: [internal-extensions] exten => 100,1,Goto(mainmenu,s,10) exten => 101,1,Dial(SIP/101,30) exten => 101,2,Voicemail(u101) exten => 101,3,Hangup() exten => 102,1,Dial(SIP/102,30) exten => 102,2,Voicemail(u102) exten => 102,3,Hangup() dialing 100 then hits "mainmenu" [mainmenu] exten => s,10,Answer exten => s,11,Wait(1) exten => s,12,Background(buddy-cloud/welcome2) exten => s,13,WaitExten(15) exten => s,14,NoOp(Number dia...
2007 Apr 26
0
problem with A400P01 OpenVox
...n make make install make samples make config Mi configuration files: zaptel.com loadzone=es defaultzone=es fxsks=1 zapata.conf [channels] signalling=fxs_ks usecallerid=yes callwaiting=no threewaycalling=no transfer=yes cancallforward=yes ; valores validos 256(32ms),512(64ms),1024(128ms) echocancel=yes echotraining=yes echocancelwhenbridged=no rxgain=0 txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming ;busydetect=yes ;busycount=10 answeronpolarityswitch=yes hanguponpolarityswitch=yes polarityonanswerdelay=600 ;callprogress=no progzone=es channel =&g...
2006 Jan 10
2
Problem with Action:Originate with ASterisk Manager
...ollowing. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel: SIP/101\r\nExten: 102\r\nPriority: 1\r\nContext: default\r\n\r\n"); send(msock, buff1, 255); But I get the following error response from Asterisk-Manager Response: Error Message: Missing action in request Later I enabled the DEBUG Log in Asterisk I can see the following - >>>>>> During...
2007 May 09
3
select menu
Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Dial(SIP/101,30,Ttm) sorry about my english, thanks to all beyyyyyyyy -------------- next part -------------- An HTML a...
2007 May 08
2
outgoing calls
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2015 Jun 15
5
Calling multiple phones at ones
Hello group! I?m new to Asterisk but got one running finally :) Now I?m trying to solve following problem. I have company Automated Attendant and each employee have SIP phone at home, SIP phone in office, cell phone. I want all those 3 phones to be ?one?. So, if someone calls our company number and dials my extension - I?d like 3 phones to ring at the same time. What is this feature and where
2007 Apr 18
43
[RFC PATCH 00/35] Xen i386 paravirtualization support
Unlike full virtualization in which the virtual machine provides the same platform interface as running natively on the hardware, paravirtualization requires modification to the guest operating system to work with the platform interface provided by the hypervisor. Xen was designed with performance in mind. Calls to the hypervisor are minimized, batched if necessary, and non-critical codepaths
2007 Apr 18
43
[RFC PATCH 00/35] Xen i386 paravirtualization support
Unlike full virtualization in which the virtual machine provides the same platform interface as running natively on the hardware, paravirtualization requires modification to the guest operating system to work with the platform interface provided by the hypervisor. Xen was designed with performance in mind. Calls to the hypervisor are minimized, batched if necessary, and non-critical codepaths
2007 Apr 18
33
[RFC PATCH 00/33] Xen i386 paravirtualization support
Unlike full virtualization in which the virtual machine provides the same platform interface as running natively on the hardware, paravirtualization requires modification to the guest operating system to work with the platform interface provided by the hypervisor. Xen was designed with performance in mind. Calls to the hypervisor are minimized, batched if necessary, and non-critical codepaths