SIP calls don't work with POTS

I have a PBX with 6 POTS lines as any CID / any DID

That Inbound route rings to a queue.

When I add a SIP inbound route and point it to an extension of a soft phone, calls can ring in and out, but there is no voice. I have 5060 and 10K-20K forwarded to the PBX.

Any suggestions? Can SIP mix with POTS?

I should mention that I am running the soft phone extension on a from-restricted context and forcing it out the SIP Trunk. I see on a tail -f /var/log/asterisk/full that it is going out.

But I get this error:

– Called sip/
[Mar 17 22:31:55] VERBOSE[25933] logger.c: – SIP/sip-08ed5f00 is making progress passing it to SIP/2000-b6836450
[Mar 17 22:31:56] WARNING[5401] chan_sip.c: Remote host can’t match request NOTIFY to call ‘MjFjMmI2ZDYyM2M4ZGNiMGI1ZjdmMzc5ZjNmOWExNjY.’. Giving up.
[Mar 17 22:32:06] VERBOSE[25933] logger.c: – SIP/sip-08ed5f00 answered SIP/2000-b6836450
[Mar 17 22:32:26] WARNING[5401] chan_sip.c: Maximum retries exceeded on transmission NDcxOGNhNTAwNjRhNzBiMWUxZGJkNTkyMTQ0YmU2MmY. for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
[Mar 17 22:32:26] WARNING[5401] chan_sip.c: Hanging up call NDcxOGNhNTAwNjRhNzBiMWUxZGJkNTkyMTQ0YmU2MmY. - no reply to our critical packet (see doc/sip-retransmit.txt).

Sounds like you got a NAT problem rather than a firewall one.

Can you make sure you define that correctly in SIP Settings in FBX?

Its important to define the internal range properly.