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Data Ceter Setup Remote Extension will not ring

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rhett

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Hello Group,

here is the setup

3CX running on windows 2003 server in data center public ip address.
dns on windows 2000 server set to resolve to sip.pbtoolkit.com
i tried adding the extra server locaton dns entry but could not enter the periods so it looks like _sip_upd and i left off the pbtoolkit.com since i could not enter the . ( i had the same troulbe with or with out hte dns entries and just using the ip address of the 3cx server)
I have 3 remote office extensions
all 3 extensions are registerd.


08:57:32.250 StratInOut::eek:nHangUp [CM104007] Call(6): Call from Ext.5502 to 5501 has been terminated by Ext.5502; cause: CANCEL; from IP:70.58.103.44
08:57:16.656 CallConf::eek:nIncoming [CM103002] Call(6): Incoming call from 5502 (Ext.5502) to sip:[email protected]

when calling from 5502 to 5501 here is the logs
the phone 5501 does not ring, same thing from any extension to any other.

i do not have any voip gateways set up yet just trying to get the basic ext to ext wroking first.

i have tried it with the remote extension checked and unchecked same results. i am not really sure what the remote extensopn check box does.

using polycom ip301 on 2 exts and linksys spa2102 on the 3rd.
all were working fine with junctionnetworks.com virtual pbx system.

What could I be doing wrong?

Thanks

Rhett
 
Data Center Setup

This version of the 3CX server software will not work in a coloation
setup unless all ip phones have public ip address. I was told that v4
will come out sometime next month & everything will be routed thru
port 80 which make the data center setup work. Most retail routers
block ports & it is very hard to figure what ports are being blocked.
The 3CX has certain ports it uses but nat's don't allways use those
same ports on both ends & their starts your problems. Most nat's
routers will pass all traffic on ports 21/25/53/67/80/443 without
any problems. Check your firewall will setup on your nat/router to
see if you can fix your problem. We are still working with different
Linksys routers to see how they work on this type of setup.

ecwilson
 
so i will need to uses a server in the colo that does not have a webserver currently ruunning on it i guess if i have to use port 80.

the same phones were working fine on junctionnetworks.com pbx setup. what ports would i have to have opened to make 3cx work. i thought the phone just opened a port to the pbx when it registerd.

do you think setting the registration time out to a low number will help like 3 mins or so to keep the port open for ring requests etc..

I am new to this and do not exactly know how all the signaling gets triggerd. does the phone continually check or does the registration process open a connecton that remains open and the pbx just sends the info to the phone.

also once a call is set up to a voip gateway or extenstion does the voice traffic flow throught the pbx server or do the phones communicate directly to each other?

Rhett
 
The 3CX server is not the problem, it is where your phones are located
& if they are not using public ip's then the router/nat has to have the
ports open in order for the phone to work with the server. Their is a list
of ports that need to be opened. Look in the forum for them.

ecwilson
 
Just as a FYI, this I know well :lol:
List of default ports used by the 3cx phonesystem:
5060 (stanard sip port) TCP/UDP
5061 (secondary sip port) TCP/UDP
5480 (portgre) (only on local machine)
5481 (apache) (only on local machine)
5482 (media server) (only on local machine)
3478 (STUN Server) TCP/UDP
7000-7500 (Default port for internall calls) (only on local machine)
9000 - 9003 (ports for external calls through VOIP Gateway) UDP
10000-20000 (RTP ports) (Used if no stun available for audio)

Link https://www.3cx.com/community/threa...is-gets-a-sticky.1697/?highlight=ports+sticky
 
Thanks I have been looking all over for that. :D

ecwilson
 
List of default ports used by the 3cx phonesystem:
5060 (stanard sip port) TCP/UDP
5061 (secondary sip port) TCP/UDP
5480 (portgre) (only on local machine)
5481 (apache) (only on local machine)
5482 (media server) (only on local machine)
3478 (STUN Server) TCP/UDP
7000-7500 (Default port for internall calls) (only on local machine)
9000 - 9003 (ports for external calls through VOIP Gateway) UDP
10000-20000 (RTP ports) (Used if no stun available for audio)

ok this is a good list i found it the other day. here is the deal

(only on local machine) i would assume is for the 3cx box and it is in the data center all ports are open and it is on a public ip address.

5060 and 5061 i would assume are opend autiomatically when the phone registerd from behind my linksys wrt router. if i were to add port mapping on the dsl router i can only map 5060 to one phone ip address and that makes no sense, as i had 4 phones working just fine with anohter pbx located in a data center all behind the same wrt and it all worked no router fddling required.

STUN is the default setting on the 3cx box i made no changes there.

here is my observations i have only been trying to figure out this viop stuff for 2 weeks so i am most likely wrong but here goes.

i believe that when the phones register with the PBX that a constant connection is set up between the pbx and the phone. i come to this conclustion as to how fast the phones rings and the watching the light blink on my switch. it is not constantly blinking so i do noit believe the phone is checking every second for a ring command but when you dial extenstion to ext the phone rings almost instantly so the PBX must have the ability to push the ring out to the phone and if it is behind a DSL router the connection must be open and waiting for a responce. that makes me believe the 5060 through the wrt router to the correct phone has to be done automatically as all calls into the DSL router will have to be mapped to the correct phones.

there may be a time out that is closing the 5060 port mapping and maybe 3600 seconds is way too long and i need to have the phone register more often to keep the port open.

i am going to try 180 seconds and see what happens and see if that works for me.

can someone tell me exactly what happens when extenstion a calls extension b and how the signaling all happens? this will help solve the problem or at least put us all on the same page.

thaks to everyone for the input so far.

Rhett
 
one more thing i gatherd by reading the forum here.

it looks like all this port forwarding talked about here is to make the pbx accessable to the internet on a public ip address and has nothing to do with the remote phone end. the remote phone end is what the stun is all about it figures out what the public ip addres of the phone is.

so my data center 3cx box ip is public and all prots are open, stun is on so it can find the public ip address of the phones.

am i looking at this backwards?

Rhett
 
Here is an update,

Ok have most of my phones working now.

here is the deal with NO PORT MAPPING required on the remote phone end.

As i thought the nat opens the hole by itself, your phone or the pbx needs to keep it open by sending a pacekt every 30 seconds or so. you can test this your self by using the 3cx soft phone or any other phone behind a remote nat. start up the phone and within 30 seconds of it registering have another extension call it and wala it works just fine. And you can talk just fine. Now hang up and wait a coule of mins and try it again it will not work.

I have the linksys SPA2102 gizmo at 2 remote locations, the spa2102 has a cool little section on line config pages called NAT keep alive jsut turn it on and set it to 30 seconds and your remote phone will work all day long. i set it to reregister every 3 mins just in case of a router reset or pbx reset. it also has a stun server setting area on the bottom of the page you can uses if you want to it worked either way.

my polycom 301 is the last hold out, i can dial the other extensions just fine but can not talk as it does not have the keep alive stuff so to make it work. i would have to give it a fixed ip address and open the ports to it. but still working on a fix as it has a nat section i am not sure how to configure but that may fix but i won't hold my breath.

after looking on the net i see some phones have the nat keep alive built in and others do not. so look for this feature.

when i was testing junction netowrks i remember them telling me they took care of the nat issues so their server must do the keep alive from the pbx end. the polycom phone worked just fine and they recommend using it.

So if 3cx will add the keep alive feature to their end all phones should work with out all this port mapping nonsense.

thats all for now.

Rhett
 
Thanks I will check this out & see if we can get out test server working
along with our phones :shock: Also if we get them working we would
also like to do a test between servers if you wish to see how that can
be setup. :)

ecwilson
 
It seems to be working :D I will be testing it for the rest of the day
& would like to do some testing with others to make sure it works. :D

ecwilson
 
Great then it is repeatable.

now i am trying to get my system working with gafachi.com for outgoing calls tehn i will sign up for a did once this is working, gafachi gives you $1.00 credit to test with befoer you even put in your credit card info!

it looks like i have to use ip authentication with them as i do not see where to put the username and password on the VOIP provider page on 3cx.

does anyopne have any experience with using this provider.

Rhett
 
It seems that the GXP's are very ticky ip phones & we now had to
play around with the nat symmetric routing to get all 4 phones to work.
But they all work now :shock: The next big problem will be have
other phones plug into the system from other cities with different
routers & firewall to see how each phone acts with each router/firewall.
I will also try to order some Linksys phones to do some testing with
them. Also I looked over the gafachi.com web site & will add them to
our list of did providers (THANKS) :) I hope v4 will end this advanced
setup stuff & make the ip phones plug/play. :wink:

ecwilson
 
ecwilson,

You might want to consider running a STUN server on the 3CX so you have dependable connectivity for your remotes. The open source Winstun server works OK when combined with SRVANY which turns the commandline program into a service. Have your own stun server ensures that your remotes won't be affected by other servers or networks.

Saw someone makes a Windows sipserver that include STUN server but it costs $50.00. Free is better.
 
Once we learn a little more about the stun software will be running our
own stun server. But for now we will be using others servers.

ecwilson
 
Our freelance tech has a softphone & it does not work from location to
location. It was setup from has office & that is the only place it will work.

ecwilson
 
Verify the extension is setup as external in 3CX
Verify the stun server used at the remote is the same one being used by the 3CX server.
Download winstun client utility and run it on a PC behind the remote firewall to verify proper operation of STUN behind a remote firewall.

Works for me.
 
Verify the extension is setup as external in 3CX.....it is 8)

Verify the stun server used at the remote is the
same one being used by the 3CX server. :?: Do you mean on the laptop

Download winstun client utility and run it on a PC behind the remote
firewall to verify proper operation of STUN behind a remote firewall.
Will do :p

Thanks
ecwilson
 
ecwilson said:
Verify the stun server used at the remote is the
same one being used by the 3CX server. :?: Do you mean on the laptop

ecwilson

Yes, in the softphone config, is the stun server name the same as the one in the 3CX server config?
 
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