what kind of firewall is this? On Oct 26, 2011 6:50 PM, "Tony Graziano" <tgrazi...@myitdepartment.net> wrote:
> > On Oct 26, 2011 6:21 PM, "Adrien Guillon" <aj.guil...@gmail.com> wrote: > > > > To address your points: > > > > > sipx server should be behind NAT. It's IP address should be using > stun or have the public address manually input. > > > > The public address has been input into Devices -> Gateway -> xxx -> > > NAT -> Public IP address > > > > > the itsp should NOT be doing nat traversal for you. > > > > I have configured their web interface to indicate I am not behind a NAT. > > > > > stop using the iptables sip conntrack modules, they will not be of > any help. just setup iptables to do symmetric nat. > > > > Done, I have removed them. > > > > > make sure your trunk say to use the public address for call setup. > > > > Not sure how to do this. > > system>server>nat > > > > Please see the attached sip log, and thanks for all of your help :-) > > A call was dropped around 18:18:53, the first call I made I tried the > > wrong extension so I disconnected myself. > > > > AJ > > > > On Wed, Oct 26, 2011 at 2:04 PM, Tony Graziano > > <tgrazi...@myitdepartment.net> wrote: > > > They have not so far, because there is a public IP showing in the FS > > > negotiation. I don't think it should be there when you are behind NAT. > I > > > checked mine and it did not do that. > > > > > > On Wed, Oct 26, 2011 at 1:59 PM, Adrien Guillon <aj.guil...@gmail.com> > > > wrote: > > >> > > >> Before we get too far into the analysis, can someone confirm that my > > >> NAT looks about right, to eliminate that issue first? > > >> > > >> AJ > > >> > > >> On Wed, Oct 26, 2011 at 11:54 AM, Tony Graziano > > >> <tgrazi...@myitdepartment.net> wrote: > > >> > it is probably more so of an issue with the way the carrier treats > > >> > reinvite. > > >> > I don't recall seeing a not allowed here in the trace files so I > don't > > >> > know > > >> > why codec is being brought up. there are multiple things wrong with > his > > >> > firewall config so maybe once that is fixed this will be easier to > work > > >> > on. > > >> > > > >> > On Oct 26, 2011 11:46 AM, "winson (Elabram)" < > winson.k...@elabram.com> > > >> > wrote: > > >> >> > > >> >> .... is it codec issue? > > >> >> > > >> >> > > >> >> On 26/10/2011 04:07, Adrien Guillon wrote: > > >> >> > Hi everyone, > > >> >> > > > >> >> > I have been working on incoming calls from a sip trunk, and > debugging > > >> >> > potential issues. Right now, calls are disconnected immediately > > >> >> > after > > >> >> > I dial an extension from the AA (when I call externally). I'm > pretty > > >> >> > sure the NAT is configured properly, and I'm starting to narrow > down > > >> >> > the problem. The NAT uses nf_conntrack_sip rather than > explicitly > > >> >> > opening RTP ports. I used tcpdump to monitor incoming calls, and > I > > >> >> > find events such as (right before disconnection): > > >> >> > > > >> >> > 19:40:25.689135 IP bm-srv-01.voicenetwork.ca> 123.456.1.12: > ICMP > > >> >> > bm-srv-01.voicenetwork.ca udp port 19222 unreachable, length 208 > > >> >> > > > >> >> > I have discussed this with a friend, and one potential issue > could be > > >> >> > how the phone network is configured. My phones are firewalled so > > >> >> > that > > >> >> > they can only communicate with the SipX server. I am not sure if > the > > >> >> > transfer negotiation is attempting to pass the connection > directly to > > >> >> > the phone, which then has no path back (and is not really > reachable > > >> >> > from the NAT system). > > >> >> > > > >> >> > Any suggestions? > > >> >> > > > >> >> > AJ > > >> >> > _______________________________________________ > > >> >> > sipx-users mailing list > > >> >> > sipx-users@list.sipfoundry.org > > >> >> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > >> >> > > > >> >> > > >> >> _______________________________________________ > > >> >> sipx-users mailing list > > >> >> sipx-users@list.sipfoundry.org > > >> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > >> > > > >> > _______________________________________________ > > >> > sipx-users mailing list > > >> > sipx-users@list.sipfoundry.org > > >> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > >> > > > >> _______________________________________________ > > >> sipx-users mailing list > > >> sipx-users@list.sipfoundry.org > > >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > > > > > > > -- > > > ====================== > > > Tony Graziano, Manager > > > Telephone: 434.984.8430 > > > sip: tgrazi...@voice.myitdepartment.net > > > Fax: 434.465.6833 > > > > > > Email: tgrazi...@myitdepartment.net > > > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > > Telephone: 434.984.8426 > > > sip: helpd...@voice.myitdepartment.net > > > > > > Helpdesk Contract Customers: > > > http://support.myitdepartment.net > > > Blog: > > > http://blog.myitdepartment.net > > > > > > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > > Ask about our Internet Fax services! > > > > > > _______________________________________________ > > > sipx-users mailing list > > > sipx-users@list.sipfoundry.org > > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > > _______________________________________________ > > sipx-users mailing list > > sipx-users@list.sipfoundry.org > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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