On 4/19/2012 3:25 PM, Simon Brûlé wrote:
How can I do a capture with wireshark on the SipXecs server?
If you google a little you will find it.
About the ALG you think that the other Router that give the DHCP to my
Laptop and the Wan adresse of my router would have the Sip ALG activate?
That would be the only thing inbetween your softphone and the sipx
server... right?
http://screenshots.portforward.com/Cisco/Linksys_E2500/Management.htm
2012/4/19 Gerald Drouillard <gerryl...@drouillard.ca
<mailto:gerryl...@drouillard.ca>>
On 4/19/2012 2:58 PM, Simon Brûlé wrote:
I added 192.168.175.0/24 <http://192.168.175.0/24> to the
intranet subnet and I still have the same problem.
2012/4/19 Gerald Drouillard <gerryl...@drouillard.ca
<mailto:gerryl...@drouillard.ca>>
On 4/19/2012 2:37 PM, Simon Brûlé wrote:
Hi, I know I already posted something very similiar to this
problem but I haven't found a solution to it so here i am
reposting my problem but with more precision this time.
I have a softphone (Jitis) on a Ubuntu 11.10 installation
connected to the network of the company.
I have a router Linksys E2500 connected to the same network.
The laptop have the adresse 192.168.175.136 giving by dhcp
and the router have the adresse 192.168.175.22 giving by
dhcp too.
On that router I have my SipXecs server and 2 hardphones
connected. My SipXecs server have the adresse 192.168.0.1,
the internal adresse of the router is 192.168.0.2 and the 2
hardphones have dhcp adresse given by the SipXecs server.
The problem is the following :
When I call with the softphone that is registered on the
SipXecs server to a hardphone that is registered on the
server too the call get there but there is no sound on
either side and the hardphone is still flashing like the
call is still coming and i didn't answer it. By the way the
phone is a Polycom 321.
When i call from the Hardphone to the softphone everything
is fine except that the softphone can't do any sound but he
can hear the hardphone.
The firewall on the SipXecs server is disabled, the firewall
on the router is disabled too, the SipXecs server is in the
DMZ of the router, Sip ALG is disabled on the router too.
On the SipXecs server System --> Internet calling I have
the Nat traversal enabled and the Server behind nat. The
intranet domain is the default one and for the intranet i
put the 192.168.0.0/24 <http://192.168.0.0/24>.
You may need to add 192.168.175.0/24
<http://192.168.175.0/24> also if it is local.
I have seen polycom phones act like this before. In my case:
The user portion of a SIP dialog MUST match the ACK and if it does
not match exactly the phone will ignore it. Without a valid ACK
the phone won't start sending RTP and the UI won't show the call
as answered. You may want to do a capture on the sipx server and
look at the results with wireshark.
Sounds like you may still have ALG at the gateway on the
192.168.175.0 network.
--
Regards
--------------------------------------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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--
Regards
--------------------------------------
Gerald Drouillard
Technology Architect
Drouillard& Associates, Inc.
http://www.Drouillard.biz
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