similar to: FW: Strange and complete failure of Asterisk 1.8 - part 2

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is

Marcelo Terres wrote: > I noticed another different behaviour. > > In older versions, when I call rasterisk, I receive some informations > about it. Fox example: > > [root at pbx2 ~]# rasterisk > Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer<markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type

Hello, Is there a way to let someone access to Asterisk CLI and type whatever command (s)he likes but the shell command (the ones started by !) ? Ideally, it could be an argument to rasterisk: rasterisk --no-shell When done, a session could be like this: > pjsip show endpoints ... > core reload ... > !rm /etc/foobar Forbidden Suggestions ? Best regards -------------- next part

Hello, I'm remotely managing an asterisk setup using an OpenVPN client on this Asterisk box, connecting to an OpenVPN server of mine). This box is mainly connected to PSTN. It is also connected to the Internet, only for remote management. The former ADSL link has recently been replaced by a new 4G link (UMTS). I'm connecting to this box from a Debian Jessie/Gnome Terminal combo. With

2016-03-03 18:08 GMT+01:00 Duncan Turnbull <duncan at e-simple.co.nz>: > > > On 4/03/2016, at 5:31 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Hello, > > > > I'm remotely managing an asterisk setup using an OpenVPN client on this > Asterisk box, connecting to an OpenVPN server of mine). > > > > This box is mainly connected to

Hello! After upgrading my production machine to 1.2.1(used to be 1.2.0) on friday i experienced strange behaviour yesterday, i received deadlock-avoided-messages and channels refusing to hangup on span1(used for inbound calls), both messages in all cases paired: Jan 9 17:40:01 WARNING[30003] chan_zap.c: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. Jan 9 17:40:01

I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it

Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried getting a coredump, but it doesn't coredump. I know there are a lot of errors in the log below, but most of those just say it'll not load a module, and no big deal. When launching from commandline (not service script) here is what happens. http://pastebin.com/3GFe6fG9 Travis Ryan Director of

Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten => TestTrakt,1,Dial(ZAP/1-2/517255333) exten => TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [TestTrakt at

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hi, everyone I'm investigating if I could give asterisk CLI access to one of our clients. If I add that user to asterisk group and set his shell to /usr/sbin/rasterisk, is there a possibility for a user to brake our of asterisk CLI to normal shell? Thanks in advance

Hi everyone. Having experimented a but with a prototype of a system I described in an earlier thread (Reading DTMF sent by callee during a SIP call), I decided to implement my requirement by transferring the call to another extension. This way, the callee can open the door by pressing #1, and the dial plan for extension 1 takes care of the rest. This works when I make a typical SIP to SIP call,

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

https://bugs.freedesktop.org/show_bug.cgi?id=94225 Bug ID: 94225 Summary: Mesa crash with "nouveau" driver and Minetest Product: Mesa Version: 10.2 Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau Assignee: nouveau

The Asterisk Development Team has announced the release of Asterisk 11.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.21.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a