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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