Hi all,
I am having an issue that appear to act like NAT but most of the traditional NAT fixes are not addressing the issue. I am using Asterisk 13 18.3 with PJSIP
The setup
Phone1 (ext 702) ->>
Phone_Server (ip: 10.10.1.47) --> Public_IP1 = Public_IP2 <-- Phone2 (ext 186)
The problem: When i make a make a call from phone 1 to the phone server everything is fine (as it should be they are on the same LAN). But when I try to make a call from phone 2 or the phone server I get no audio. I Suspect the issue is related to the registration as when things dont go through firewall1 they look better and work
Non working config
uri user_agent qualify_timeout authenticate_qualify via_addr via_port endpoint
sip:702@10.10.1.25:5060 Aastra 6867i/4.2.0.2023 3 no 10.10.1.25 5060 702
sip:186@Public_IP2:32658^3Brinstance=ffccbaf40665d202 X-Lite release 5.1.0 stamp 89322 3 no 192.168.43.82 61946 186
Working config
uri user_agent qualify_timeout authenticate_qualify via_addr via_port endpoint
sip:702@10.10.1.25:5060 Aastra 6867i/4.2.0.2023 3 no 10.10.1.25 5060 702
sip:186@Public_IP2:32658^3Brinstance=ffccbaf40665d202 X-Lite release 5.1.0 stamp 89322 3 no Public_ip2 61946 186
My Transport appears as:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.10.1.0/22
local_net=127.0.0.1/32
allow_reload=yes
external_media_address=Public_IP1
external_signaling_address=Public_IP1
My Endpoints: (both are the same)
Endpoint: 186 Not in use 0 of inf
InAuth: 186/186
Aor: 186 10
Contact: 186/sip:186@public_ip2:32658;rinstance 1aa255fd26 Unknown nan
Transport: transport-udp udp 0 0 0.0.0.0:5060
ParameterName : ParameterValue
100rel : yes
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 186
asymmetric_rtp_codec : false
auth : 186
bind_rtp_to_media_address : false
call_group :
callerid :
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : from-Phone
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username
inband_progress : false
incoming_mwi_mailbox :
language :
mailboxes :
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : false
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth :
outbound_proxy :
pickup_group :
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
subscribe_context :
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : transport-udp
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
Here is my registration on the no working conneciton
[0K<— Received SIP request (542 bytes) from UDP:Public_IP2:3467 —>
REGISTER sip:Public_IP1 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.82:50770;branch=z9hG4bK-524287-1—9f5ddc59455c5674;rport
Max-Forwards: 70
Contact: sip:186@192.168.43.82:50770;rinstance=825588bfbf119967
To: "Test_lap1"sip:186@Public_IP1
From: "Test_lap1"sip:186@Public_IP1;tag=0816683b
Call-ID: 89322OTI0OGQ3MjVmYjdiZmJiNmY1MTg0ZGFlNTczNzhlOTE
CSeq: 1 REGISTER
Expires: 3600
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
User-Agent: X-Lite release 5.1.0 stamp 89322
Content-Length: 0
[0K<— Transmitting SIP response (553 bytes) to UDP:Public_IP2:3467 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.43.82:50770;rport=3467;received=Public_IP2;branch=z9hG4bK-524287-1—9f5ddc59455c5674
Call-ID: 89322OTI0OGQ3MjVmYjdiZmJiNmY1MTg0ZGFlNTczNzhlOTE
From: “Test_lap1” sip:186@Public_IP1;tag=0816683b
To: “Test_lap1” sip:186@Public_IP1;tag=z9hG4bK-524287-1—9f5ddc59455c5674
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1513710457/f607d1ed2c2d9f4972c4fb3c51789aa1”,opaque=“3421a4040440bac6”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 13.18.3
Content-Length: 0
Here is my registration on the Working one
[0K<— Received SIP request (817 bytes) from UDP:Public_IP2:61694 —>
REGISTER sip:Public_IP1 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.82:58806;branch=z9hG4bK-524287-1—15a95a509a801b58;rport
Max-Forwards: 70
Contact: sip:186@Public_IP2:61694;rinstance=bc00805d342416a7;expires=0
To: "Test_lap"sip:186@Public_IP1
From: "Test_lap"sip:186@Public_IP1;tag=9ae0f434
Call-ID: 89322OGRkNjI4YTFlMDljYzBlYjAwZTQyOTY5MjNjNDFmN2E
CSeq: 5 REGISTER
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
User-Agent: X-Lite release 5.1.0 stamp 89322
Authorization: Digest username=“186”,realm=“asterisk”,nonce=“1513710159/9c11e2611bf72d48765daf897b4368cb”,uri=“sip:Public_IP1”,response=“702d6e75b4d961bc30f2c99c94ffd3ea”,cnonce=“7e81db88f1ee64a004f7cc4d03465fbc”,nc=00000004,qop=auth,algorithm=md5,opaque=“7ae2b2f6275085a3”
Content-Length: 0
[0K<— Transmitting SIP response (565 bytes) to UDP:Public_IP2:61694 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.43.82:58806;rport=61694;received=Public_IP2;branch=z9hG4bK-524287-1—15a95a509a801b58
Call-ID: 89322OGRkNjI4YTFlMDljYzBlYjAwZTQyOTY5MjNjNDFmN2E
From: “Test_lap” sip:186@Public_IP1;tag=9ae0f434
To: “Test_lap” sip:186@Public_IP1;tag=z9hG4bK-524287-1—15a95a509a801b58
CSeq: 5 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1513710243/8284c715aa37d14d13584111873dd552”,opaque=“53c5d9300aaeb178”,stale=true,algorithm=md5,qop=“auth”
Server: Asterisk PBX 13.17.2
Content-Length: 0
I would prefer to have things behind the firewall so any assistance would be appreciated