No calls to peers outside of local network (Asterisk 13)


#1

I am trying to setup a phone outside of the local network and have run into problem where asterisk cant place calls to them.
What i am getting is the good old “Retransmission timeout reached on transmission”

Now i have configured the external ip and hostname and the phone can registers normally.
If a call placed from external phone through Asterisk either to local or external landline phone all work as expected BUT when calling the phone from Asterisk then phone never rings and i get re transmission error.

Now i noticed this output when placing a call Strict RTP learning after remote address set to: 192.168.0.2:60290
This has phone local address instead of remote, is that correct ?

I can provide any configuration and sip debug output if required.
Thanks.

Global Settings:

UDP Bindaddress: 192.168.1.205:45070
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX
SDP Session Name: Asterisk PBX 13.24.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain: :45070
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network Settings:

SIP address remapping: Enabled using externhost
Externhost: :45070
Externaddr: :45070
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0

Peers Settings

[oleg]
insecure=invite
type = friend
defaultuser = oleg
secret =
host = dynamic
context = allcalls
callgroup = 1
pickupgroup = 1
qualify=yes
disallow = all
allow = all

Debug output
pbx*CLI> sip set debug peer oleg
SIP Debugging Enabled for IP: 178.59.20.224

<— SIP read from UDP:178.59.20.224:52902 —>
INVITE sip:46@EXTERNAL HOST NAME:45070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:52902;branch=z9hG4bK-524287-1—75dce11ef907102e;rport
Max-Forwards: 70
Contact: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
To: <sip:46@EXTERNAL HOST NAME:45070>
From: “Oleg”<sip:oleg@EXTERNAL HOST NAME:45070>;tag=d36d737d
Call-ID: 94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU
CSeq: 1 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 332

v=0
o=- 13193858940822279 1 IN IP4 192.168.0.2
s=X-Lite release 5.4.0 stamp 94388
c=IN IP4 192.168.0.2
t=0 0
m=audio 62280 RTP/AVP 9 8 120 0 84 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:84 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
— (13 headers 12 lines) —
Sending to 178.59.20.224:52902 (NAT)
Sending to 178.59.20.224:52902 (NAT)
Using INVITE request as basis request - 94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU
Found peer ‘oleg’ for ‘oleg’ from 178.59.20.224:52902
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 120
Found RTP audio format 0
Found RTP audio format 84
Found RTP audio format 101
Found audio description format opus for ID 120
Found audio description format speex for ID 84
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|alaw|g722|speex16|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|speex|g722|opus)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7ff684014c20 – Strict RTP learning after remote address set to: 192.168.0.2:62280
Peer audio RTP is at port 192.168.0.2:62280
Looking for 46 in allcalls (domain EXTERNAL HOST NAME)
sip_route_dump: route/path hop: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239

<— Transmitting (NAT) to 178.59.20.224:52902 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:52902;branch=z9hG4bK-524287-1—75dce11ef907102e;received=178.59.20.224;rport=52902
From: “Oleg”<sip:oleg@EXTERNAL HOST NAME:45070>;tag=d36d737d
To: <sip:46@EXTERNAL HOST NAME:45070>
Call-ID: 94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:46@EXTERNAL ADDRESS:45070>
Content-Length: 0

<------------>
– Executing [46@allcalls:1] Dial(“SIP/oleg-0000000e”, “SIP/oleg”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14900
Adding codec ulaw to SDP
Adding codec g723 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


-- Called SIP/oleg

Retransmitting #1 (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


Retransmitting #2 (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


Retransmitting #3 (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


Retransmitting #4 (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


Retransmitting #5 (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


Retransmitting #6 (NAT) to 178.59.20.224:52902:
INVITE sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK5ba7942f;rport
Max-Forwards: 70
From: “Oleg” <sip:oleg@EXTERNAL HOST NAME:45070>;tag=as651e2b58
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:oleg@EXTERNAL ADDRESS:45070>
Call-ID: 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 881

v=0
o=root 1472840116 1472840116 IN IP4 EXTERNAL ADDRESS
s=Asterisk PBX 13.24.1
c=IN IP4 EXTERNAL ADDRESS
t=0 0
m=audio 14900 RTP/AVP 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


[Feb 5 18:27:36] WARNING[8194]: chan_sip.c:4069 retrans_pkt: Retransmission timeout reached on transmission 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb 5 18:27:36] WARNING[8194]: chan_sip.c:4093 retrans_pkt: Hanging up call 694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [46@allcalls:2] Hangup(“SIP/oleg-0000000e”, “”) in new stack
== Spawn extension (allcalls, 46, 2) exited non-zero on ‘SIP/oleg-0000000e’
Scheduling destruction of SIP dialog ‘94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU’ in 6400 ms (Method: INVITE)

<— Reliably Transmitting (NAT) to 178.59.20.224:52902 —>
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.0.2:52902;branch=z9hG4bK-524287-1—75dce11ef907102e;received=178.59.20.224;rport=52902
From: “Oleg”<sip:oleg@EXTERNAL HOST NAME:45070>;tag=d36d737d
To: <sip:46@EXTERNAL HOST NAME:45070>;tag=as6dbf79c9
Call-ID: 94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘694d0749049b7139255e13a85597b208@EXTERNAL HOST NAME’ Method: INVITE

<— SIP read from UDP:178.59.20.224:52902 —>
ACK sip:46@EXTERNAL HOST NAME:45070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:52902;branch=z9hG4bK-524287-1—75dce11ef907102e;rport
Max-Forwards: 70
To: <sip:46@EXTERNAL HOST NAME:45070>;tag=as6dbf79c9
From: “Oleg”<sip:oleg@EXTERNAL HOST NAME:45070>;tag=d36d737d
Call-ID: 94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘94388NWJiNzRkYzQ5MTEzY2RjZGU0OGQyYjQ4MzE2MzJjZGU’ Method: ACK

<— SIP read from UDP:178.59.20.224:52902 —>

<------------->
Reliably Transmitting (NAT) to 178.59.20.224:52902:
OPTIONS sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239 SIP/2.0
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK69a48f84;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@EXTERNAL HOST NAME:45070>;tag=as7e6878df
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239
Contact: <sip:asterisk@EXTERNAL ADDRESS:45070>
Call-ID: 68c3c73c7f3f44d04ff3001b52960318@EXTERNAL HOST NAME
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 05 Feb 2019 16:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:178.59.20.224:52902 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP EXTERNAL ADDRESS:45070;branch=z9hG4bK69a48f84;rport=45070
Contact: sip:192.168.0.2:52902
To: sip:oleg@178.59.20.224:52902;rinstance=659e1c8834871239;tag=7fe47266
From: “asterisk” <sip:asterisk@EXTERNAL HOST NAME:45070>;tag=as7e6878df
Call-ID: 68c3c73c7f3f44d04ff3001b52960318@EXTERNAL HOST NAME
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Allow-Events: talk, hold
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘68c3c73c7f3f44d04ff3001b52960318@EXTERNAL HOST NAME’ Method: OPTIONS
pbx*CLI>


#2

I’d suggest providing the actual configuration and the SIP trace (sip set debug on) of an attempt.


#3

I have updated the question with more info.
What i am doing here is calling same extension from same phone,the effect is the same if call is places from other extension.
I also replaced the host names and ip addresses in the output and configuration.