There doesn’t seem to be any RTP from the .136 device in this capture. Have you check at that device to see if it is sending RTP, is there a Firewall blocking RTP from that direction?
From: [email protected] [mailto:[email protected]] On Behalf Of Simon Brûlé Sent: Thursday, April 19, 2012 1:21 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Nat Problem I used the following command on the SipXecs server tcpdump -n -s 0 -i any -w filename.cap and then I transfer it on my computer so I could open it with my Wireshark and have a look at it. I joined the file to this e-mail so you could take a look and tell me what you think. This one is from a call I did from the softphone to the Hardphone where this one bugged like I described earlier. Thanks. 2012/4/19 Simon Brûlé <[email protected]> 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 list [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/
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