[asterisk-users] asterisk and openvpn and sip

[Giorgio Incantalupo]

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.
>>     
>
>   


-- 
Giorgio Incantalupo, mailto:gincantalupo at fgasoftware.com
Voice at work - The Agile PBX http://www.voiceatwork.eu
FG&A srl - http://www.fgasoftware.com
Tel: 02 997663.14, Fax: 02 91390172