search for: vvvvvvr

...nd (and optional background) color to specified. A full list of colors may be found in include/asterisk/term.h* But nowhere could I find what format the color code should be. I have tried all possible permutations, none of them works: ASTERISK_PROMPT="%Cn[COLOR_BLUE] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[32;128] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[32;] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[;32] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[;COLOR_CYAN] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[32|128] %H:...

...Can the behavior described above be configured ? 2) I would like to be able to change the color of the asterisk prompt as described here: https://www.voip-info.org/asterisk-cli-prompt/ I tried all possible permutations, but none work: ASTERISK_PROMPT="%Cn[COLOR_BLUE] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[32;128] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[32;] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[;32] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[;COLOR_CYAN] %H: " asterisk -vvvvvvr ASTERISK_PROMPT="%Cn[32|128] %H: " ast...

It seems on my phone, which is hooked up to a large pbx network powered by an asterisk server, that it will randomly start ringing with a callerid# of 2013 which is its username for that phone. I have looked and been watching on the asterisk command line with the -vvvvvvr switch and nothing has been seen that indicates a reason for this random ringing. This leads me to think that this trouble is not involving the asterisk server, however I am bothered by it and would like to find out what is causing the trouble. I was wondering, does anyone here have any ideas...

Thanks Anthony. I did it on the server, according to https://www.voip-info.org/wiki/view/port+forwarding However after doing it, when running Asterisk I get the following message sudo asterisk -vvvvvvr No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to '/' for compatibility. How and where can it be set? My server ifconfig: lo Link encap:Local Loopback inet...

Hi * Users, What is the way from the dial-plan to get the IP address of an incoming or outgoing SIP call? I can see the IP address of the SIP call using 'sip show peers' from the CLI. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

...XXXXXX,2,Macro(fastbusy) ; [local] ; ; Master context for local, toll-free, and iaxtel calls only ; include => parkedcalls include => trunklocal include => iconnectlocal [extensions] ; include => local =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-= [root@lizzie asterisk]# asterisk -vvvvvvr == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD-05/28/04-09:54:40, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ======================================================================== = Connected to Asterisk CVS-HEAD-05/28/04-09:5...

We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in the following manner: Current Setup: Telco-> T1->PBX Desired Setup: Telco-> T1-> Asterisk-> T1-> PBX. I am first trying to setup the Asterisk -> T1->PBX part without disturbing existing setup so I can get asterisk to forward calls to PBX and once that is done, I would try to move the telco

Hi, At my house, I have two POTS lines. Both are connected to my * server on a TDM400P card. As an example, say the phone numbers are (919)555-1212 and (919)555-1213. I also have four SIP extensions, an ATA with a fax machine, and a DID coming in from an ITSP. I have an autoattendant configured that talks and allows users to forward to the extension they choose, but my family doesn't like

Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers Now to Antony questions: On 06/06/2017 04:36 PM, Antony Stone wrote: > On Tuesday 06 June 2017 15:18:32 andre castro wrote: > >> I just installed asterisk in a debian server. >> All seems to be running fine, but the

...Dial(SIP/${ARG1},120,tr) exten => s,n,Congestion() When I try dialling 912345678 the above configuration thinks that I am dialling a SIP number, and does not bridge the call to the ZAP FXO channel. What am I doing wrong please ? This is the debug output :- Code: mailhub asterisk # asterisk -vvvvvvr == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.15, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show war...

...RetryTime: 60 WaitTime: 30 Context: services Extension: 0001932485457 Priority: 1 SetVar: MSG=hello cp test.call /var/spool/asterisk/outgoing/ && chown asterisk.asterisk /var/spool/asterisk/outgoing/test.call && chmod 777 /var/spool/asterisk/outgoing/test.call && asterisk -vvvvvvr Asterisk 1.6.2.9-rc2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public Li...

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -