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
<[email protected]> 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)" <[email protected]>
> 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
>> > _______________________________________________
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>> > [email protected]
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>> >
>>
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