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/ >
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