Retransmission timeout reached on transmission in Asterisk

Question

Asterisk

Hello,
There was a problem of the following nature:
When I call, for example, from extension 100 (I connect to it from the Internet) to 101 (which is inside the network), the conversation goes on, and after about 6 seconds an error occurs in the Asterisk logs:

[Jan 30 08:04:58] WARNING[2119]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission 52IXBJSBuDga-b7X7wM2-OHXdPZOnxw1 for seqno 8311 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Jan 30 08:04:58] WARNING[2119]: chan_sip.c:4288 retrans_pkt: Hanging up call 52IXBJSBuDga-b7X7wM2-OHXdPZOnxw1 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Asterisk looks to the Internet through a router. Outward forwarded ports sip 5060 (tcp/udp) and rtp 10000-20000 (udp).
rtp.conf

[general]
rtpstart = 10000
rtpend = 20000

sip.conf

[general]
tcpenable = yes
disallow = all
allow = alaw,ulaw

[phones](!)
type = friend
context = phones
host = dynamic
nat = no
qualify = yes

[100](phones)
defaultuser = 100
secret = *****

[101](phones)
defaultuser = 101
secret = *****

Tell me where to dig and how to be, I honestly read https://wiki.asterisk.org/wiki/display/AST/SIP+Ret... but I didn’t find anything intelligible about my case.
CentOS 6, Asterisk 12

Answer 1

If Asterisk is behind nat, the localnet and externip parameters must be specified. In your case, the endpoint behind the nat receives 200 OK when responding, but most likely the phone tries to send an ACK to it to the Contact address announced by Asterisk: sip:[email protected]:5060, which it can reach due to NAT, of course , can not. As a result, the timer fires and Asterisk breaks the call, because ACK to 200 OK is required.
That is, either include externip=external router ip so that asterisk specifies this address in requests and responses to addresses other than those specified in localnet, or on clients, enable the use and sending of the rport option, which tells the client, when choosing a signaling address, to focus on the data of the rport parameter, and not the parameters of the SIP message.

Answer 2

Try specifying externip and localnet in sip.conf , and also enable nat=yes in the settings of the peer that connects through the external.

Answer 3

why do you have extras 100 and 101 in the condition, but in the log I see [email protected] ??
Found peer '101' for '101' from 86.102.40.95:62312 -- maybe another error in the condition?
Via: SIP/2.0/UDP 192.168.9.113 :62312;branch=z9h****;received=86.102.40.95;rport=62312 -- where did this address come from?
Pier 101 what IP??
Draw the network more carefully