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 <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!
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