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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