1.6 VS. 2.0

yukon9

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#1
I useSIP trunk from a local VOIP provider. The trunk registered well on both elastix server.

Elastix 2.0 performs very good, but with a small bug, can't add event on agenda.

However, when I use elastix 1.6, SIP trunk and extension register well. call out well, but always problem with incoming call, always " the number you dial is not in service,........" Anything wrong?
 

coryjsanders

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#2
Check your sip_nat.conf file and make sure you have this:
nat=yes
externip=<your public ip>
localnet=192.168.1.0/255.255.255.0 <this is mine. You may be a 10.10.x.x/24 or something else.>
 

DaveD

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#3
ss-no service is inbound route issue

Try anyDID/anyCID for that inbound trunk to see if it works
 

danardf

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#4
Hi.

Remember that Elastix V2.0 isn't a stable version, is only a RC version!!!
So, we can't compare the performances with the 1.6, but only the packages. :huh:

Regards
 

dicko

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#5
Hey Franck:

Sorry I missed your "meeting"
 

yukon9

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#6
coryjsanders said:
Check your sip_nat.conf file and make sure you have this:
nat=yes
externip=<your public ip>
localnet=192.168.1.0/255.255.255.0 <this is mine. You may be a 10.10.x.x/24 or something else.>

I did it in both ELASTIX 1.6 AND 2.0
 

yukon9

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#7
danardf said:
Hi.

Remember that Elastix V2.0 isn't a stable version, is only a RC version!!!
So, we can't compare the performances with the 1.6, but only the packages. :huh:

Regards
It's interesting I didn't encounter big problem with elastix 2.0 till now, but lots of problem with 1.6.
 

yukon9

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#8
DaveD said:
ss-no service is inbound route issue

Try anyDID/anyCID for that inbound trunk to see if it works
tried, still doesn't work. perhaps it's my VOIP provider's problem, but it seems no problem when I install it on elastix 2.0.
 

yukon9

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#9
I also try to use elastix 2.0's extension as elastix 1.6's SIP strunk, encounter same.
 

DaveD

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#10
If you have allow anonymous sip set to yes in general settings does it work?

If it does update freepbx to 2.6 and turn off allow anonymous sip and should work if trunk is configured correct
 

danardf

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#11
yukon9 said:
It's interesting I didn't encounter big problem with elastix 2.0 till now, but lots of problem with 1.6.
It's false, you can't say this. :huh:
Each installs are different. It's possible that you haven't any problem with your config, but it's not really the case for the others.

The 1.6 is more stable than the 2.0.
Since how many time a RC is more stable than the stable version? :blink:

Every Alpha, Beta, and RC versions are make to be tested, So.

Myself, I use the 1.6.21 and no problem. Every extensions and trunks works fine. ;)

With the 2.0 Beta, without Freepbx 2.7, this version not works.
The RC is better than the Beta, ouf course, but thas all.

You can't install this version in production. But, you can install it for any test and reports bugs that the Dev Team try to fix every bugs reported.
 

danardf

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#12
dicko said:
Hey Franck:

Sorry I missed your "meeting"
Irrelevant
Hey Dicko! How are you.
I forgive you Dicko.

You can download the pdf file to see my work. Here
;)
 

Telco

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#13
I have found that some SIP providers require that your pbx be in the DMZ or on the outside network. Some don't work behind a nat. They will register one way but not both. Check to see that both your pbx's are on the outside or in a DMZ. Just a thought.
 

yukon9

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#14
Contact my VOIP provider and follow their intruction on sip trunk setting. The problem is resolved finally.


Very appreciated for everyone herein. :)
 

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