similar to: Colour coding the dialplan -- NoOp and ANSI codes?

You can post process the logs with something like sed. See: https://superuser.com/questions/380772/removing-ansi-color-codes-from-text-stream On 7/23/20 5:10 PM, Andrew Yager wrote: > Hi, > > Is there a way to drop the ANSI colour strings from log files? In > particular, I've got JSON logging throwing logs over to ES, but they > have the ANSI colour escape sequences. >

Have you tried starting asterisk with the "-n" param? -n Disable console colorization On Thu, Jul 23, 2020 at 5:11 PM Andrew Yager <andrew at rwts.com.au> wrote: > Hi, > > Is there a way to drop the ANSI colour strings from log files? In > particular, I've got JSON logging throwing logs over to ES, but they have > the ANSI colour escape

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Hi, Is there a way to drop the ANSI colour strings from log files? In particular, I've got JSON logging throwing logs over to ES, but they have the ANSI colour escape sequences. Ideally I don't want to lose coloured logs from the console though, and I can't "see" a way to do this. Ast 16 at the moment… Andrew -------------- next part -------------- An HTML attachment was

Hi Mark, Are you still having audio problems between outside SIP channels? Make sure that you have set the following for all SIP channels in your sip.conf canreinvite=no -- sudhir > Message: 2 > Date: Mon, 26 Jul 2004 22:46:22 -0400 > From: Leif Madsen <leif.madsen@gmail.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Nat...again.... >

Thanks for your reply. canreinvite has been set to "no" from the beginning...still no luck. Maybe I'll be able to take a trace of it tonight...we'll see...but any thoughts at all are appreciated! -Mark > > Hi Mark, > > Are you still having audio problems between outside SIP channels? Make > sure that you have set the following for all SIP channels in your

Hi Leif, Thanks for the information. I checked the /tmp/ folder and there was core #### files and I tried to back trace it but it was not showing the cause of that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from past few days its going on fine. I have also researched and found that version 1.4.17/18.1 had the issue of channel stuck up as well as random asterisk crashes.

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

Emrah wrote: >> This is an asterisk-users question, and would have been more appropriate to have >> asked there. >> >> Instead of setting up your conferences in meetme.conf, you could set them up >> dynamically in the dialplan, and then you can control whether the user is >> prompted for a pin or not when joining the conference, based on whatever logic

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a medium sized business that is interested in implementing * as their PBX system. They currently have a Panasonic system with Panasonic handsets that they are going to replace Asterisk with, as the current system is maxed out, and they don't even have voicemail capabilities. I have been considering using an Adtran Atlas 550 with FXO and

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 OK, so I have a local extension that a phone can call to take it to voicemail. I don't want it to exit out to a fast busy tone, as I would rather it allow the user to simply continue on and call a new number (without having to physically release the line first). The [intern] context is where everything goes by default (sip.conf for example has

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am using a TDM40B, and have managed to compile mpg123 and turned on MOH. Problem I am having is that it is choppy, staticy, and sounds like white noise pretty much. I have search the archives to see if this problem had been resolved, but I haven't found anything yet. Has anyone had this problem and resolved it? I am calling from

Greetings Asterisk Users, I'm happy to announce that Russell Bryant and Leif Madsen have volunteered to host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon central time. Russell and Leif are project leaders and have collaborated on two Asterisk books: Asterisk: The Definitive Guide and Asterisk Cookbook , both published by O'Reilly & Associates. Asterisk: The

Hi: Which was released for free download under a Creative Commons license for "The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen". Regards. --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go