Elastix / FreePBX Configuration

[Madman88]

This thread is GOLD

Thanks guys

[mh348]

I get this from the console:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

According to some sites it could be the codec or the port forwarding... I installed the licensed g729 which now works for outbound calls..

I also forwarded the ports, udp 10000:10500 and udp 4569, but still no change..


Also another thing, for the pstn trunk (telkom), there is a delay when it rings, the calling party first hears three rings, only then does my fone start ringing.. I added the line "immediate=yes" in the chan_dahdi.conf so it starts ringing immediately, but still only rings after 3 rings on the remote side..

If I want my phones to ring immediately, in which custom files can I add the "immediate=yes".. Currently I have it in the chan_dahdi.conf but it still only starts ringing after 3 rings..

[Gnome]

What makes you say IAX is insecure?

Actually, you know what, I checked again. They've fixed the security problems since I last checked. Ignore that comment

I have been using FreePBX for a while and it works well. You can still add your own custom code using the *_custom.conf files or if you need to hack extensions.conf directly there is a freepxb_override.conf file. For the day-to-day configs FreePBX is great.

Ah, Awesome! Thanks! I can't work without at least some access to the dial plan.

[morkhans]

Actually, you know what, I checked again. They've fixed the security problems since I last checked. Ignore that comment

hehe, cool

Ah, Awesome! Thanks! I can't work without at least some access to the dial plan.

Once you understand how FreePBX touches the dial-plan it's actually quite easy to work with it when you need to make customizations.

[landfiets]

Don't forget to make a sip_nat.conf file in Elastix so your external IP adress can be related to your server. Without doing this it can cause oneway audio and bad NAT communication.
Also take care of your codecs. When in the provider details you use G729 but in your extension details you force to use something else, it might go wrong as well, even with transfering a call.
So in your extension you can edit: disallow=all & allow=alaw. In your provider do the same. Or if you only want to use G729 do the same for that one. Or allow=alaw&G729
Using the right codecs is often the problem in getting circuits all busy messages.
I know almost everything about Elastix, so ask if you want. Always use type=friend in provider and never user or something else. That way you can use the same configuration for outgoing and incoming without 2 times filling out the details in the provider tab. So only fill it out once, and leave the other. Register string is almost always with any provider: usernameassword@provider/username

[nic777]

Hi guys,

Great thread.

I am in the process of setting up Elastix with FNB connect as well and having been following this thread in detail today.

I noticed there is some skipped parts though, which I assume is things that were resolved but not reported back.

I am now in the same situation where if I try to dial my FNB number, it goes straight to the "The user at 087.. is not available" and get the lines in the console saying:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

May I ask how you resolved this? Was it trunk settings or the firewall?

Many Thanks
Nick

[nic777]

I get this from the console:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

According to some sites it could be the codec or the port forwarding... I installed the licensed g729 which now works for outbound calls..

I also forwarded the ports, udp 10000:10500 and udp 4569, but still no change..

Hi,

I am also in the same process of setting up Elastix with FNB Connect and now stuck in this same position where the phones don't ring.

It looks like as I read down the thread, you were able to solve this. May I ask how you got incoming calls to come in through FNB Connect?

Many Thanks
Nick

[mh348]

Hi,

I am also in the same process of setting up Elastix with FNB Connect and now stuck in this same position where the phones don't ring.

It looks like as I read down the thread, you were able to solve this. May I ask how you got incoming calls to come in through FNB Connect?

Many Thanks
Nick

Hi... I still haven't resolved these issues

I have 2 or 3 sip extensions setup, and purchased only 1 g729 codec license, maybe if i remove the other sip extensions and just leave 1 it might work..

I've been a bit lazy and haven't worked on my pbx for some time now.. Other issues I have is that there is still a delay on the Incoming call, the caller hears the phone ringing about 3 times, then only does it start to ring on my side..

Another issue I have is that I cannot transfer external calls to any of my extensions... So if I pickup a call on extension '1' I can't transfer the call to any other extension..

I've since switched back to the old system as I had new cables installed for my elastix pbx, and currently just using the elastix pbx for internal calls..

[nic777]

Hi... I still haven't resolved these issues

I have 2 or 3 sip extensions setup, and purchased only 1 g729 codec license, maybe if i remove the other sip extensions and just leave 1 it might work..

I've been a bit lazy and haven't worked on my pbx for some time now.. Other issues I have is that there is still a delay on the Incoming call, the caller hears the phone ringing about 3 times, then only does it start to ring on my side..

Another issue I have is that I cannot transfer external calls to any of my extensions... So if I pickup a call on extension '1' I can't transfer the call to any other extension..

I've since switched back to the old system as I had new cables installed for my elastix pbx, and currently just using the elastix pbx for internal calls..

Oh hectic

Perhaps then someone can assist both of us then

I am also about to contact FNB Connect whether they can assist, especially since last month they have started to offer a business package.

Looks like we both stuck on the trunk incoming settings, not the incoming ROUTE which I believe to be different?

Some blog mentioned "you need a 'default' context for incoming when using FNB" but don't know what exactly he means by that.

[MrGreen]

Has anyone used Elastix/Asterisk with VoIP provider Dataphone from ICTGlobe (South Africa)?

I can make calls out to cell phone numbers. Sometimes to Telkom numbers, but never to 0800 or 0860 numbers.
I would like to route all outgoing calls through my VoIP provider.

Incoming call do work from my VoIP provider.

Thanks

[Madman88]

Ok kinda OT, but all of a sudden none of my trunks are showing in the fop.
I have done nothing to the pbx in the last bit, tho I have considered upgrading to the latest Elastix from 1.6.

For all my searching none of the solutions I have found work. I would go ahead with the upgrade but apparently v2 has a similar problem.
Anyone have any ideas?

[mh348]

Ok kinda OT, but all of a sudden none of my trunks are showing in the fop.
I have done nothing to the pbx in the last bit, tho I have considered upgrading to the latest Elastix from 1.6.

For all my searching none of the solutions I have found work. I would go ahead with the upgrade but apparently v2 has a similar problem.
Anyone have any ideas?

Hi

Did you come right with your setup.. It's almost a year now that my elastix box has been switched off... I want to sort it out so I can start using it...