Hello good afternoon; I’ve been having problems connecting to my asterisk server at home from outside. So I decided to create a VPN tunnel using openvpn. I’m using xlite softphone client to connect. I’m able to connect using the VPN tunnel and I can call my other extensions at home but I can not hear them and they Can’t hear me eather.
Can anyone help me out with this?
any suggestions please.
thanks!
Mm
please post your extension.conf setup and any relevant CLI output for the call. Also please confirm all required RTP ports are open between the sip phone and the asterisk server.
Hello; this is the config files I have…
I’m using Ext 1000 to connect from outside. I already have open ports and I am able to place calls internally only. If I’m connecting through my VPN why not act the same as If I was internally? do you think I need to add a route or there is a traffic issue?
thanks for your help.
sip.conf
[general]
externhost=Mysite.home.com
localnet=192.168.1.0/255.255.255.0
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0
[1000]
type=friend
context=internal
host=dynamic
secret=*******
canreinvite=no
[1001]
type=friend
context=internal
host=dynamic
secret=******
canreinvite=no
extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[internal]
exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
exten => 1001,1,Verbose(1|Extension 1001)
exten => 1001,n,Dial(SIP/1001,30)
exten => 1001,n,Hangup()
[phones]
include => internal
has anyone here used “OpenVPN” with asterisk to avoid problems with NAT?
I am using 2 SIP phones behind NAT and I have no problems.
- Is this SIP client on laptop or desktop? If it is on laptop - can you make test if it could connect and have no problems at your work?
- I used VPN but it was VPN over SSH - and we had no problems.
There are more easy ways to solve NAT problems. By the way - I had only bad experience with SIP phones, except SIP phone for Android
And of course you must set NAT=yes
If [1000] is at home, you must set in sip.conf
NAT=yes
PS
Last is required if phone is behind NAT, in case you use VPN - you must check VPN address - you can have lot of “funny” situations if SIP phone listens on other address
Last thing is dump of traffic at server - to see where it sends UDP packages, and if it receives UDP packages from your SIP client.
Do you have a SIP phone in a remote location behind NAT too? and no problems?
Do you recommend putting asterisk on a public IP without a firewall?
I think I have a network issue at my house…I guess I can try hooking up the asterisk directly to my DSL line and see if I can connect using the WAN IP. If I do this test do I have to foward the ports on my job firewall as well or it should not matter?
thanks!
Mm
Whatever else you do…NEVER EVER do that!
My configuration:
SIP Phone <-> router(with firewall) <->Internet <-> Asterisk server (with public IP and firewall)
I had hard times - no voice in one or both directions for some of phones. Then I started dumping packages on router and on Asterisk - to see which leg is damaged. Only this way I fixed problems with phones. I set STUN on one of phone - lot of iterations until solved problem.
The VPN is good because it helps the UDP audio stream transmit through the cloud.
SIP is sometimes funky over routers, and even VPNs.
What type of router/routers is it? I have seen a lot of those junky dsl ‘truck roll’ modems be the culprit as well.
-Jake
www.voipcitadel.com
[quote=“voipcitadel.com”]The VPN is good because it helps the UDP audio stream transmit through the cloud.
SIP is sometimes funky over routers, and even VPNs.
What type of router/routers is it? I have seen a lot of those junky dsl ‘truck roll’ modems be the culprit as well.
-Jake
voipcitadel.com[/quote]
I had two routers and it was causing a conflict with asterisk. I took one out for now and my sip clients are conecting now.
Thanks all for the help