Why does a call from a local SIP client to a remote one not work?

Question

Asterisk

Greetings!
There is a hardware SIP server BAS-IP SIP-PBX-16 based on Elastix connects to a Mikrotik router with a dedicated IP, forwarded UDP ports 5060 and from 10000 to 20000.
Inside the network, everything works fine SIP clients call each other.
If one client is online and the other is remote, then the call to the remote client from the internal network does not go through at all.
If a remote client calls a client located inside the network with a SIP server, then everything is OK, voice and video are transmitted without problems.
If both SIP clients are remote, then calls also do not go.
In all these cases that remote clients that internal are correctly registered on automatic telephone exchange.
What I did:
1.registered in sip_general_custom.conf

externip=10.20.30.40
localnet=192.168.1.0/255.255.255.0
nat=yes
canreinvite=no
registertimeout=20
registerattempts=0
maxexpiry=3600
minexpiry=60

2. forwarded ports 5060.

add action=netmap chain=dstnat dst-port=5060 in-interface=ether1 protocol=udp 
to-addresses=192.168.1.99 to-ports=5060

3. forwarded ports 10000-20000

add action=netmap chain=dstnat dst-port=10000-20000 in-interface=ether1 protocol=udp 
to-addresses=192.168.1.99 to-ports=10000-20000

Errors that asterisk gives when calling from a local SIP client to a remote one.

[2016-11-26 14:50:17] WARNING[2898][C-000000a1]: chan_sip.c:11130 process_sdp_a_audio: Got Opus useinbandfec=1
[2016-11-26 14:50:17] WARNING[2898] [C-000000a1]: chan_sip.c:11130 process_sdp_a_audio: Got Opus useinbandfec=1
[2016-11-26 14:50:25] WARNING[2898]: chan_sip.c:4030 retrans_pkt: Retransmission timeout reached on 51dbc32e450ac01d10bc6 .30.40:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Ret...
Packet timed out after 8064ms with no response
[2016-11-26 14:50:25] WARNING[2898]: chan_sip.c:4030 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- Seehttps://wiki.asterisk.org/wiki/display/AST/SIP+Ret...
Packet timed out after 8064ms with no response
[2016-11-26 14:50:25] WARNING[2898]: chan_sip.c :4059 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Ret...
[2016-11- 26 14:50:25] WARNING[2898]: chan_sip.c:4059 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/ display/AST/SIP+Ret...

What am I doing wrong?

Answer 1

Put the allowed codecs in Asterisk g729 (if any), ulaw, alaw
And the same ones on the client software
You have errors related to the Opus codec. If it works with others - pick it.

Answer 2

on mikrotik to start
/ip firewall service-port disable sip