Idea for a forum.

rslrdx

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#1
Hi everyone!

I was setting up a new elastix server for a small company that uses two different sip phone carriers, and noticed that the biggest problem I had, was finding the information needed to setup elastix tp connect to this carriers, not that they didn't provide the information to use an asterisk server, but they seem to give only the information as if you were running your own custom asterisk install. This information is (usually) not formated to match the elastix GUI, so it can be a little confusing if you are not used to setting up asterisk servers.

Based on this I have a suggestion that might just help other people and also make it easier to spread elastix to others. I think that if we had a forum just to keep the sip providers connection configuration information as it should be entered in the elastix GUI, it would be of great help... I myself spent at least 4 days trying to match the information to setup 2 sip providers.

I know there is the
 

ramoncio

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#2
I think this is a really nice idea.
Here is my config for justvoip:

Peer config:
allow=g729&alaw&ulaw&gsm&g723
disallow=all
host=sip.justvoip.com
secret=mypassword
type=peer
username=myuser

Register string:
myuser:mypassword@sip.justvoip.com:5060
 

wiseoldowl

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#3
Before anyone tries reinventing the wheel, you may want to visit this page on the FreePBX site first:
Howto: Setting up VOIP Provider Trunks

A few general points...

With many commercial providers you do NOT put anything in the user context or user settings. That is because they are treating you as an extension, not a peer. This is explained more fully in How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail):

..... note that with [certain] providers, you may have to move that context statement from the USER details to the PEER details section. This is why calls from some SIP providers sometimes fail to come in at all - they effectively never "see" the User context and details, therefore they don't see the context statement there and have nowhere to go. It's also why you sometimes see instructions for sip providers that leave the User context and User details sections totally blank, but include a context statement in the peer details - in most such cases it's because the provider is treating the customer as an end user (like someone using a softphone or a VoiP adapter) rather than as a peer, and they aren't sending DID information.
You MUST have a context= statement (usually context=from-trunk), and if you're not using the user section then that context statement must go in the peer details.

For incoming calls to route properly, make sure your registration string is in this format:

accountid:yourpassword@your.provider/yourDIDnumber (NO SPACES even though the forum software adds one!) :angry:

Note the /yourDIDnumber at the end - many providers don't suggest it but it can make all the difference whether your inbound routes work. Failing that, again I'd refer you to How to get the DID of a SIP trunk when the provider doesn't send it (and why some incoming SIP calls fail).

If you have codec related statements (disallow and allow) make sure the disallow statement(s) (particularly disallow=all) come before the allow statements.

There are probably other things I could mention but can't think of them offhand. :huh:
 

dicko

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#4
ramoncio,


quote:
allow=g729&alaw&ulaw&gsm&g723
disallow=all

is about to break, I believe newer versions of asterisk will allow/disallow sequentially, thus you will end up having no codecs to use.

reference:-

http://www.asterisk.org/doxygen/1.2/Config_sip.html


snippet from above url

;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
 

ramoncio

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#5
You are right dicko.
I've been experiencing problems lately with the codecs, and I noticed that removing the line disallow=all solved the problem.
First disallow, then allow.
In olders versions of Asterisk and Elastix it worked fine, I think before 1.3 it worked.
 

rslrdx

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#6
Well, this is not going to reinvent the well, but would be nice to have it here.

While wiseoldowl is right, we don't need to reinvent something that is already working, there is nothing wrong with having it here either. Elastix was not the first asterisk based
 

Chilling_Silence

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#7
I dont think its specifically a bad idea, but two things come to mind:
1) There's tons of carriers, many international, but what about the more local ones such as for me here, in New Zealand?
2) How they might go out-of-date, or be 'incomplete' ...

That said, having a button somewhere on the Elastix WebGUI that links to a page on the Wiki wouldnt specifically be a bad idea :D
 

ramoncio

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#8
It would be even nicer to make a Elastix module where you select your provider from a list (at least the most common providers, maybe clasified by country?), input your username/password, and the trunk is automatically configured in freePBX.
And I think it would't be very hard to accomplish.
 

rafael

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#9

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