Asterisk 1.6 commands

siptellnet

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#1
Hi
I did a new and clean installation of elastix 2.0
Somebody knows why the "restart now" and "show translations" doesn't work in elastix 2.0 ??
I installed the codec g723 and g729, but the trunks no register well.
I can make oubound calls but can't receive.
FREEPBX shows trunk register, DASHBOARD show NO trunk register
Any idea please
SIPTELLNET
 

danardf

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#2
Hi.

Please, read some doc before write this!!

Asterisk is changed from 1.4 to 1.6.
Since 1.4, when you put your cmd line, you could see more informations after your cmd that said for example:

The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.

Now, for restart now, you must put: core restart now
I installed the codec g723 and g729, but the trunks no register well.
It's not a codec issue!

I can make oubound calls but can't receive.
Read the doc, Elastix Without Tears!

FREEPBX shows trunk register, DASHBOARD show NO trunk register
Of course!

Any idea please
Yes, read the doc Elastix Without Tears, Try to find (by Search Forum) the solution into the Elastix Forum.
You have enough information into the Elastix forum to fix your problem. ;)

Here, you haven't displayed anything. no config trunk, no info about your operator. nothing!!

That you said is so enormous. :silly:

Verify your trunk parameters with your operator.
For every test, don't use the g729/723 codec for the moment.
Until you have the trunk problem, don't use it.
When your trunk will be up, then you could put your preferred codecs.

Regards
 

dicko

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#3
As suggested by danardf, please read either the documentation or remember the last time you tried

restart now

it told you that it should be

core restart . . .


you are using commands that no longer work in Asterisk 1.6, please try at the asterisk CLI

help

most are now

core show . . .
and

core restart . . . .
etc.



to to re-educate yourself to the new terminology, which I might add has worked for a while and you were warned a few years ago that the old ways would be abrogated in 1.6
.


you can take a horse to water (for two years now) but you can't make it drink . . .

dicko
 

siptellnet

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#4
I used elastix 1.5/1.6 for 2 years with this trunks, same server, same network, same router, same trunk configuration, same SIP provider.
Only I changed to elastix 2.0, 32 bits.
Ok, the "core" is the new command, thanks
My mistake was using codec for asterisk 1.6 instead for asterisk 1.6.2.
Now, I see the "core show translation" ,but even the trunk does not work well.
I continue reading
 

dicko

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#5
You should have said you were using stolen codecs, the legitimate ones that come with Elastix 2.0/Asterisk 1.6 work perfectly well. Nevermind the ethics, you apparently sell the service in the U.S., (from your first two posts/spams on these fora) so are without doubt "not legal" with Digium, who own the rights to g729.
 

siptellnet

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#6
For all the people OPEN MIND
http://asterisk.hosting.lv/
G.729 and G.723.1 codecs x86 (and x86_64) Linux and FreeBSD binaries for Asterisk open source PBX
DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm.
This site is very help full for testing and learning
I am not the owner, I am only a final user, and like to share information

Eventually I unistalled the codec g723 and g729 were unnecesary
I fix the problem with the trunk, the VOIP provider has an issue with the TRUNK NAME and USER CONTEXT firts letter.
After thar the trunk register well

Good luck

A reflexion: "if you don't have something nice or motivating to say, please keep in silence"
 

dicko

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#7
Perhaps you "duck and dive", perhaps not. Quoting from your first post

. . . Send an email to admin@xxxxxxxx, and I will provide a new SIP Trunk or Virtual Number of 7 digits, in few minutes.

I note your use of "first person".

As to motivation:-

http://www.voip-info.org/wiki/view/Aste ... +Licensing

for anyone who cares to jeopardize a business venture for the sake of 10 bucks a channel lifetime use.
 

Centar

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#8
Doesn't the licensing for the g729 codec depend on weather your transcoding or not?

For example if your incoming is g729 and your outgoing is g729 then your not doing any transcoding of codecs and there are no licensing requirements.
You only need to pay for the license if you will be transcoding g729 into another codec.

This was the way I read and understood the "rules" for Digitum on their g729 licensing requirements.

Please correct me if I am wrong.
 

dicko

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#9
That is quite right, If you have phones which carry there own license they will work, what won't work is MusicOnHold, VoiceMail, IVR's or Conferences. If you don't need these then don't buy the licenses.

dicko
 

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