[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[David Villasmil]

Check out what IPs are offered in the SDPs in asterisk. Make sure they’re
both public IPs.
If you only have 1 asterisk, forwarding the rtp port range configured in
asterisk from the firewall to asterisk should do it.


On Thu, 14 Jan 2021 at 08:23, Mark Allen <mark at allenclan.co.uk> wrote:

> Thanks Adrian
>
> The firewall has SIP-ALG disabled and just forwards ports from externally
> to where they need to be internally - so ports 5060 and 10000 - 65535 of
> 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box)
>
> On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu <ag at ag-projects.com> wrote:
>
>> Google search for SIP ALG problem to see if this is relevant for your
>> case.
>>
>> Regards,
>> Adrian
>>
>>
>> On 13 Jan 2021, at 13:08, Mark Allen <mark at allenclan.co.uk> wrote:
>>
>> Hi all - I've been banging my head against this but not succeeding.
>>
>> Our setup...
>>
>> UAC           192.168.x.x
>>   |
>> Router        5.x.x.x
>>   |
>> (internet)
>>   |
>> Firewall      46.x.x.x maps
>>   |           directly to
>> OpenSIPS      192.168.x.x      Mid-registrar
>>   |
>> Asterisk      192.168.x.x
>>
>>
>> Current situation:
>> - UAC can register on Asterisk via OpenSIPS
>> - UAC can call destination registered on Asterisk on local n/w to
>> Asterisk box
>> - Destination extension rings and can pick up call
>> - There is no audio either way & call drops after about 30 secs (Asterisk
>> kills call with "Requested channel not available" because not RTP
>> traffic is reaching destination)
>>
>> I have tried passing audio through Mediaproxy on OpenSIPS box but with no
>> success. Using Wireshark I can see RTP traffic initiated at both ends, but
>> it doesn't reach the other end either way.
>>
>> Is there some definitive guide to setting this up correctly or are there
>> specific steps that I need to follow?
>>
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>>
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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