As i dig more and more in Wireshark i came to the conclusion that the Wireshark information that I just sent you is pretty much useless as I now see it. I will keep looking for some piece of information that could help.
Thanks. 2012/4/19 Simon Brûlé <[email protected]> > My Computer is connected in the Lan of the company and my E2500 is > connected in this Lan too. My Server SipXecs and my hardphone are on the > E2500. > > So there is an other router between my computer and the router that have > my Server connected on it. > > For the Wireshark part when I answer the phonecall I do from the softphone > to my hardĥone those request are coming in until i close the call. > > 109 19.596310 192.168.175.22 192.168.175.136 SIP/SDP 1466 Status: 200 OK, > with session description > > Status-Code: 200 > [resent packet : True] > [Suspected resend of frame:104] > [Request Frame : 57] > [Response Time (ms): 10950] > > followed by this one: > > 110 19.598495 192.168.175.136 192.168.0.1 SIP 714 Request: ACK > sip:[email protected]:5060;transport=tcp > > Request-Line: ACK sip:[email protected]:5060;transport=tcp SIP/2.0 > Method: ACK > Request-URI: sip:[email protected]:5060;transport=tcp > [Resent Packet: False] > [Request Frame: 105] > [Response Time (ms): 512] > > > All those test have been done on Wireshark on the Computer with the > Softphone on it. And the 192.168.0.253 that you see is the hardphone IP > adresse. > > > 2012/4/19 Gerald Drouillard <[email protected]> > >> 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 <[email protected]> >> >>> On 4/19/2012 2:58 PM, Simon Brûlé wrote: >>> >>> I added 192.168.175.0/24 to the intranet subnet and I still have the >>> same problem. >>> >>> 2012/4/19 Gerald Drouillard <[email protected]> >>> >>>> 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. >>>> >>>> You may need to add 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 >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> _______________________________________________ >> sipx-users mailing [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> -- >> Regards >> -------------------------------------- >> Gerald Drouillard >> Technology Architect >> Drouillard & Associates, Inc.http://www.Drouillard.biz >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > >
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
