[asterisk-users] asterisk and openvpn and sip

[Darrick Hartman (lists)]

Giorgio,

tcpdump and wireshark are your friends.  Instead of guessing, capture a 
call with tcpdump then look at it with wireshark.

Darrick

On 06/18/2009 08:58 AM, Giorgio Incantalupo wrote:
> Hi Darrick,
>
> I always set canreinvite=no 'cause it gives a lot of problems if set to
> yes (and the default is).
> I made a call with rtp debug on and I noticed that normally, on the
> asterisk CLI, I see one packet sent corresponding to one packet  got
> (made a test with a local call on our production server). On the other
> server with the vpn, I get a bunch of sent followed by a group of
> got...there is something in the way the RTP packets are sent/received by
> Asterisk and maybe it can be correlated to the missing audio.
>
> Giorgio
>
> Darrick Hartman (lists) wrote:
>> Do you have 'canreinvite=no' in your sip.conf entry for this phone?  If
>> not, you should.
>>
>> On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
>>
>>> Hi John,
>>>
>>> I already have the ccd dir with the iroute (mandatory for routing to
>>> pc/phone connected to vpn client). During the last test I could register
>>> and  make a call but voice disappears after 1, 2 seconds. I'm trying to
>>> understand if it is a bandwidth problem. At the moment I have my phone
>>> connected to the openvpn client (which is its gateway) but I have to use
>>> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip
>>> (192.168.1.12) is not working. I suppose it is a  sip protocol problem:
>>> I had to change the sip.conf setting nat=yes to make the phone dial and
>>> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
>>> I keep on working on the vpn since it seems so little is missing to have
>>> a clear conversation. Let me know if your tests are successfull.