I’m having problems with asterisk and I am afraid my experience is low. I have finally installed asterisk on Debian squeeze by using the apt-get install command. The issue I’m having is that I can not use ls in asterisk to list any files or directories as described in several tutorials, does this mean with the advancement of asterisk this has been removed as a function or am I missing a module or component that I can load or install.
I am trying to teach myself asterisk by using the system and watching tutorial videos but I cannot get passed this point. Also every command I need to enter such as stop now all need to have th prefix core? Is this correct.
I would appreciate any help with this so I can finally learn dial plans and sip.conf to enable a soft phone on my system.
thank you for your non constructive advice and if i must say and on this morning after taking a break and clearing my mind, i realised my complete error and am stopping this thread purely because i realised yes ls is Linux command and no it does not work in asterisk as it is an pbx software. see - crawled and now staring to walk. i came to this forum for advice not critisim, there is enough of that in this world especially the networking and linux one. i shall indeed research my query’s should i have any more (although asterisk is pretty simple when sleep has been had) thoroughly before asking so as not to offend the very sensitive.
[quote=“dejanst”]Sorry to upset you. My intetion was not to insult you, but to guide you to a book that will answer the majority of your questions.
And I still do not know why you want to use ls in Asterisk CLI …[/quote]
as i said it was a moment of confusion and once the fog cleared ls was not an asterisk command as you so rightly pointed out. what s done is done lets move on.
i do have a genuine question which is bugging me. i have started asterisk from scratch via debian and have installed asterisk 1.6.xx i am following a video tutorial and am now at the part regarding sip.conf i have entered the same detail ie:
i then in asterisk sip reload but nothing and the same is if i iax2 reload or even just reload, i use sip show user but have no extensions (users) shown. now the tutorial offers no forum or help aspect hence my coming here. i have followed the examples to the letter but nothing. can you spot any reason as to why this is not working and i have no user 100?
i would be very grateful for any help you maybe able to offer
1.6.2.x is almost a month past final end of life, and 1.6.1.x and 1.6.0.x are over a year beyond final end of life. They should not be used for new installs, unless there are very specific requirements and you have the skills needed to back port subsequent fixes.
If you change sip.conf, and then run sip reload, and get no output, that suggests that your sip.conf is not where your installation expect it to be. For a source installs, that would be /etc/asterisk.
For security, type should be peer, not fried. nat seems to serve no useful purpose here.
(Also for security, you should use something like the MAC address, rather than the associated extension number.)
Your context name implies an untrusted caller, but the short name suggests a trusted one.
Thank you david55 sip.conf is within the source list /etc/asterisk yet as you say no output on sip reload. The asterisk 1.6 xx was the apt-get install via Debian. Should I move this and install and configure the build myself for the latest version? Would this cure my constant problems? Which is would you recommend, I have Ubuntu and fedora as well as debian.