Asterisk with openVPN tunnels: Only echo test functions - VoIP General, VoIP News, VoIP Providers, VoIP hardware, VoIP software, VoIP Technology

Hello dear forum municipality!

I stand with a buddy for the moment before a strange problem.
We wanted to draw our own SIP server up and over openVPN tunnels. The openVPN server runs so far perfectly. It runs on FRITZ! Box 2170 with my buddy.
Now we wanted to make evenly that with Asterisk. I installed the Asterisk server with me and configured he him (because I am been versed zero with Asterisk, but nothing does to the thing). We use both as Softphone Ekiga. Oh, is still mentioned: With the two computers it acts over ubuntu 8.10-Systeme.
Anyhow both interlocutors can call the echo test service and hear we us there also (thus everyone for itself). Only as soon as I try it to call or in reverse, no more clay/tone comes out. With Ekiga there is a codec announcement. This flickers only (thus codec stands briefly there and leaves then again). Also the expenditure for clay/tone and - admission-indicate flickers only.
Only why?
If it can call from its PC with the echo test, communication must function through openVPN to Asterisk. Only why don't we hear ourselves then?

And around those possibly. to answer emerging question already in advance: In the openVPN configuration Client ton client communication is permitted. I can be connected also to him by VNC and in reverse. And a network play with “Wormux” goes also. (Does not have however necessarily which with the topic and/or. To do problem

)

Have you an idea that to lie can?
About Tipps, pieces of advice and solutions or also speculations I would be pleased much!

Thank you already in advance!

Tschss,
more stefbeer

In order to help you further, you should the two sip.conf and extensions.conf posts

Hello!

So, here the two files of Asterisk:

sip.conf
Code:

[general]
 binding haven = 5061
 binding ADDR = 0.0.0.0
 LANGUAGE = de
 externip = 192.168.200.3
 external host = 192.168.200.3
 externrefresh = 10
 localnet = 192.168.200.3/255.255.255.0
 allow=g729
 allow=alaw
 allow=ulaw
 allow=ilbc
 allow=gsm
 allow=g726

 [2000]
 type = friend
 secret = ***
 host = dynamic

 [2001]
 type = friend
 secret = ***
 host = dynamic

extensions.conf
Code:

[default]
 exten => _200X, 1, set (ZIELNR=$ {EXTEN})
 exten => _200X, n, dial (SIP/$ {EXTEN}, 20)
 exten => _200X, n, Goto (s-$ {DIAL STATUS}, 1) 
exten => S-NOANSWER, 1, VoiceMail ($ {ZIELNR}, u) 
exten => S-BUSY, 1, VoiceMail ($ {ZIELNR}, b)     
exten => S-ANSWER, 1, Hangup ()    

exten => 2998,1, Answer ()
 exten => 2998, n, echo ()
 exten => 2998, n, Playback (vm-goodbye)

As said, I know myself not thereby out. But I hope it can which thereby begin

Thank you again!

Tschss,
more stefbeer

Sorry, which I must say however that in such a way am all garbage

Quotation:

[general]
binding haven = 5061
binding ADDR = 0.0.0.0
LANGUAGE = de
externip = 192.168.200.3
external host = 192.168.200.3
externrefresh = 10
localnet = 192.168.200.3/255.255.255.0

binding haven = 5061; the Asterisk is it no matter how you come purely
externip + external host = 192.168.200.3, that is an internal IP makes that for sense? - You have surely no OpenVpn in the internal LAN

Go once again everything step by step duch with Betateilchesn Asterisk course

If you Asterisk 1,4 use: external host and externrefresh are experimental.
I would try it times to remove these 4 lines from your [general] section sip.conf:
Code:

...
externip = 192.168.200.3
 external host = 192.168.200.3
 externrefresh = 10
 localnet = 192.168.200.3/255.255.255.0
…

to the topic openVPN: routest or does bridgest does you? If you indicate externip, all packages are treated such as NAT, which come from IP addresses, which “localnet” not with is aforementioned (the line with can thereby several “localnet” paints occurs). you need these parameters however only, if between the Asterisk and any Client (Phone od. other Asterisk) NAT is.

Greetings,
Laureen

Hello mario2006, hello laureen!

Thank you for your answers!

To me wrong does that I only announces itself now, but we do not have ourselves now longer time with it concerned.

@mario2006:
My buddy took over in such a way configs, because it functioned to in-house in such a way at that time with him. Only evenly over openVPN not.
But thank you for tap with the course! After guidance and a few parameter for us made times exact adapted, and now it functions.

My assumption is that it because of the NAT attitudes was. I it everywhere deactivated. Nothing must be converted, since everything runs only over the openVPN IPs.

@laureen:
We routes. How I before already said, I think strongly that it because of NAT was. Strange to say it functions since I NAT deactivated.

I could not begin much at the beginning with your information, since I knew nothing at all about NAT. But after short investigations it was then the crucial taps nevertheless.

Thank you!

Here also times thank-beautifully at beta particles for the very informative Asterisk course. I configured so far only times the bases, the remainder (like Voicemail and so) follow however still.

Tschss,
more stefbeer