those messages are normal. On Wed, Oct 13, 2010 at 10:06 AM, Scott Richesson < [email protected]> wrote:
> I've solved the problem with SIP URI dialing by switching the server > external IP discovery mechanism from STUN to a fixed IP address. I'm not > sure why this made a difference, because the media relay logs were showing > the proper external IP before I made the change. Good enough. > > Also, I noticed that I am getting about 10MB a day in my proxy log with the > following gibberish. Should I be concerned? Everything is working just > fine. CPU utilization is negligible. > > Thanks, > > Scott Richesson > Cincinnati Fan > > "2010-10-13T14:01:00.179200Z":38686:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:00.202332Z":38687:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:03.901788Z":38688:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:03.932952Z":38689:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:05.658933Z":38690:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:05.686610Z":38691:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:05.799614Z":38692:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:05.839339Z":38693:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:14.103552Z":38694:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:14.125307Z":38695:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:18.699638Z":38696:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:18.711536Z":38697:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:19.442592Z":38698:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:19.467166Z":38699:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:19.710526Z":38700:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:19.733055Z":38701:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:19.753995Z":38702:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:19.776572Z":38703:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:20.695798Z":38704:HTTP:ERR:sip.cinfan1.com:AsynchMediaRelayRequestSender-12:B68FFB90:SipXProxy:"HttpMessage::get[4] > Receiving failed on persistent connection on try 0" > "2010-10-13T14:01:22.094397Z":38705:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:22.117590Z":38706:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:22.154746Z":38707:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:22.177391Z":38708:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:22.230081Z":38709:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:22.266404Z":38710:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:23.678881Z":38711:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:23.701458Z":38712:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:25.951416Z":38713:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:25.972801Z":38714:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:27.429682Z":38715:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:27.452324Z":38716:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:27.564179Z":38717:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:27.590614Z":38718:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:31.281701Z":38719:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:31.306156Z":38720:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:31.418464Z":38721:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > "2010-10-13T14:01:31.446555Z":38722:SIP:ERR:sip.cinfan1.com:SipRouter-11:B64BBB90:SipXProxy:"SipUserAgent::send > outgoing call 1" > > -----Original Message----- > From: Tony Graziano [mailto:[email protected]] > Sent: Tuesday, October 12, 2010 4:43 PM > To: [email protected] > Subject: Re: [sipx-users] SIP URI Dialing > > When you described the ip alias getting you further it makes me point to > subnet and registration... > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: Tony Graziano <[email protected]> > To: [email protected] <[email protected]> > Sent: Tue Oct 12 16:27:53 2010 > Subject: Re: [sipx-users] SIP URI Dialing > > If its a softphone make sure srtun, ice is off and it it sending its local > address. > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: Tony Graziano <[email protected]> > To: [email protected] <[email protected]> > Sent: Tue Oct 12 16:26:17 2010 > Subject: Re: [sipx-users] SIP URI Dialing > > Make sure the intranet subnets page lists any networks that are inside your > lan and don't go through nat. Also make sure the registration say "nat" or > "no-nat" for the user in question. What is the user agent? > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: [email protected] > <[email protected]> > To: Discussion list for users of sipXecs software < > [email protected]> > Sent: Tue Oct 12 16:22:30 2010 > Subject: Re: [sipx-users] SIP URI Dialing > > OK-- I made some progress. > > Under System/Domain, I had the server IP address in there as a domain > alias. > After taking this out, things have changed for the better, but it is still > not working. > > 1) The media is now being directed to the SIPX server for relaying and NAT > compensation. Great! > 2) However, the server does NOT relay the media! > > I have a packet capture of the traffic in/out of the SIPX server if anyone > is interested. The capture shows the RTP going into sipx, but not being > relayed out. > > Any ideas on how to debug this? > > Scott > > -----Original Message----- > From: Tony Graziano [mailto:[email protected]] > Sent: Tuesday, October 12, 2010 12:08 PM > To: [email protected] > Subject: Re: [sipx-users] SIP URI Dialing > > Correct. You don't supply a sbc if sipxbridge is oing that. So that's > correct. > > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: [email protected] > <[email protected]> > To: Discussion list for users of sipXecs software < > [email protected]> > Sent: Tue Oct 12 11:53:38 2010 > Subject: Re: [sipx-users] SIP URI Dialing > > >You will either need to use sipxbridge and > Ahhhhh... But how do I do that? On the "Internet Calling" page, there is a > drop down for "Default SBC". Sipxbridge is not in the list. The only > choice is "none". > > >configure your firewall to allow the ports needed > I do have sipxbridge working fine through our firewall with a couple of > different ITSP's. > > Thanks for the reply, > > Scott. > > > From: Tony Graziano [mailto:[email protected]] > Sent: Tuesday, October 12, 2010 8:38 AM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] SIP URI Dialing > > Internet calling (uri dialing) is the same as trunking in that it needs a > SBC in order to facilitate a transaction over NAT. > > You will either need to use sipxbridge and configure your firewall to allow > the ports needed or use an SBC and configure sipx to allow internet calling > via the external SBC. > > It sounds as though your firewall is either trying to proxy the media or is > not configured for it at all if you are using sipxbridge. > > > On Tue, Oct 12, 2010 at 8:33 AM, Scott Richesson < > [email protected]> wrote: > --- Repost: Sorry if you've already seen this message --- > > Hello, > > We are using sipx 4.2.1 and have sipxbridge working fine for a couple of > outbound trunks. Our SIPX server and phones are behind a NAT. > > I have been playing around with SIP URI dialing, and I am not sure how the > media relaying/NAT busting is supposed to work. > > In my experimenting, I can get a call to connect, but I get no RTP. Upon > looking further, it looks as if the outgoing RTP is being directed directly > from the phone to the remote URI with no relay. (I would rather the RTP be > relayed so that I don't have to open up my firewall for the phones). I am > receiving no incoming RTP at all, which suggests to me that there is no NAT > compensation being done. > > Anyway, I don't have any immediate need for URI dialing, but am just > curious if this is supposed to work, or if there is any way to configure the > behavior here. > > Thank You, > > Scott Richesson > IT Manager > Cincinnati Fan > _______________________________________________ > 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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
