Hi Tony,
Thanks for the help.
1. Im registering a static IP to the ITSP (our external one)
2. they are sending calls to 5060, and im natting them to 6080 internally
3. Im not sure looks like from a tcpdump i am registering on UDP port
5060 to them, and 62941 locally.
>From then, i am contacting the ITSP on port 5060 and they are contacting me
on UDP 62941.
Im guessing this is because i have registered with them on that port, and
they are then responding to me via that port, with my firewall keeping the
state of that port.
I am using pfsense 1.2.3, which shouldn't be a problem?
Regards
Will
On Wed, Jul 20, 2011 at 2:46 PM, Tony Graziano <[email protected]
> wrote:
> Are you registering or using a static IP to the ITSP? Are they sending the
> calls to port 5080 or 5060? If you register does the provider allow you to
> confirm ip and port they see you registering on?
>
> It would appear (and you can confirm) that the call is coming in to port
> 5060 from the ITSP in the initial invite on inbound calls.
>
> Is you NAT static port NAT (symmetric ports) or dynamic? What is the
> firewall?
> On Wed, Jul 20, 2011 at 9:43 AM, William Salt <[email protected]>wrote:
>
>> Hi All,
>> I am new to SipXecs, but I have worked with a couple of voip
>> pbx's before. Ive got a few days to set up a PBX in our office, and so far i
>> am struggling to find a solution to the problem i am experiencing. So id
>> really appreciate it if someone would reach out and help me.
>>
>> I have a sip trunk from aql.com, and have configured it, it seems to work
>> (mostly)
>> I can make outbound calls to mobiles, landlines etc fine, and audio and
>> sessions establish and terminate fine.
>> However, i cannot complete inbound calls, the internal phone will ring,
>> and upon answering the call will terminate, however, the external callers
>> call will connect and stay connected untill they hang up.
>>
>> I have my firewall + NAT rules configured correctly (ports external
>> 5080-5081 = 5060-5601 internally + 30000-31000)
>> i have configured a dialplan from the username of the VOIP provider to the
>> internal extension.
>>
>> I have wireshark dumps, and a snapshot if anyone would have a look at
>> these, and possibly help diagnose my problems, i would
>> greatly appreciate this.
>>
>> Regards
>> Will
>>
>>
>> _______________________________________________
>> sipx-users mailing list
>> [email protected]
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
>
> --
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>
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