Pjsip endpoint registers fine but no response to INVITE

Asterisk V13.23.0 running on Debian 10

Actual addresses removed

I have PJSIP endpoint 110000052162169 configured to register to a remote server sip.example.com hosted on the internet. Registrations work fine.

The AOR is configured to qualify every 25 seconds, these are successful, the remote server responds with a 200.

The remote server sends SIP OPTIONS to my Asterisk server every 15 seconds, Asterisk matches these to the endpoint and responds with a 200.

Inbound calls to this endpoint are always successful, no exceptions, with two way audio.

My issue is with outbound calls, the SIP INVITE is sent by my Asterisk server however i never receive any response. I’m not really sure where to look next. Any help appreciated.

See my pjsip configuration below.

Thanks in advance




[transport_udp_5062]
type=transport
protocol=udp
async_operations=1
bind=192.168.54.235:5062
local_net=192.168.54.0/24
external_signaling_address=31.49.45.191
external_media_address=31.49.45.191
external_signaling_port=5062

[test_SIP_identify](!)
type=identify
match=sip.example.com
match=130.61.163.34
match=130.61.163.32
srv_lookups=yes

[test_SIP_registration](!)
type=registration
transport=transport_udp_5062
retry_interval=30
forbidden_retry_interval=600
expiration=300
auth_rejection_permanent=no
server_uri=
client_uri=
outbound_auth=
max_retries=500000
outbound_proxy=sip:sip.example.com:5299\;lr


[test_SIP_endpoint](!)
type=endpoint
transport=transport_udp_5062
outbound_proxy=sip:sip.example.com:5299\;lr
disallow=all
allow=alaw,ulaw,gsm
context=external-incoming
direct_media=no
incoming_mwi_mailbox=123@123


[test_SIP_aor](!)
type=aor
max_contacts=1
outbound_proxy=sip:sip.example.com:5299\;lr
qualify_frequency=25


[test_SIP_auth](!)
type=auth
auth_type=userpass



; >>---- Extension Pattern 110000052162169 ----<<
[110000052162169](test_SIP_registration)
server_uri=sip:sip.example.com
client_uri=sip:110000052162169@sip.example.com
outbound_auth=110000052162169
contact_user=110000052162169
[110000052162169](test_SIP_identify)
endpoint=110000052162169
[110000052162169](test_SIP_auth)
username=110000052162169
password=123456789
[110000052162169](test_SIP_endpoint)
aors=110000052162169
outbound_auth=110000052162169
from_user=110000052162169
from_domain=sip.example.com
[110000052162169](test_SIP_aor)
contact=sip:110000052162169@sip.example.com 

Here is the resulting SIP INVITE to which i receive no response

INVITE sip:305051@sip.example.com SIP/2.0
Via: SIP/2.0/UDP 31.49.45.191:5062;rport;branch=z9hG4bKPj3e473e74-0e8d-4904-85ef-5959008a2f64
From: "Anonymous" <sip:110000052162169@sip.example.com>;tag=1014db77-a3ab-4112-ab32-41996c00a7de
To: <sip:305051@sip.example.com>
Contact: <sip:110000052162169@31.49.45.191:5062>
Call-ID: 4680b9b4-33fb-499d-a8c0-064423116160
CSeq: 31452 INVITE
Route: <sip:sip.example.com:5299;lr>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, NOTIFY, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk Server
Content-Type: application/sdp
Content-Length:   282

v=0
o=- 666304191 666304191 IN IP4 31.49.45.191
s=Asterisk
c=IN IP4 31.49.45.191
t=0 0
m=audio 24774 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Here is an example of an incoming INVITE from the remote server

2021/06/24 15:01:10.729262 217.163.57.33:5299 -> 192.168.54.235:5062
INVITE sip:110000052162169@31.49.45.191:20127 SIP/2.0
X-PeerId: 254282427-3833532070-818759264@MSX53.gammatelecom.com
m: <sip:+447777777777@185.173.48.67:5471>
User-Agent: Intraswitch/SP16.2.2195 SCF-7 Engine/0.2.8
CSeq: 103631 INVITE
Alert-Info: <http://x>;info=alert-external
f: <sip:+447777777777@185.173.48.67:5471>;tag=8070e537-7f72-bb1e-3ccf-0a1bef3d2f47
Requested-By: sip:+447777777777@185.173.48.67:5471
Max-Forwards: 69
v: SIP/2.0/UDP 217.163.57.33:5299;branch=z9hG4bK9c9cd078475444afc0d44957e2cc3789,SIP/2.0/UDP 185.173.48.67:5471;branch=z9hG4bKe57a6b9e-6f89-9
2-defb-6e8ec61ecb73
i: 95830b75-3819-f519-0133-86d21923c930@185.173.48.67
X-Attributes: *;customer-id=0012J00002IZSH9QAP;dnis=442476549189
Remote-Party-ID: <sip:07777777777@185.173.48.67:5471>;privacy=off;party=calling;id-type=subscriber
t: <sip:110000052162169@31.49.45.191:20127>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,SUBSCRIBE,INFO
Record-Route: <sip:217.163.57.33:5299;lr>
c: application/sdp
l: 240

v=0
o=MSX53 290089985 1624543271 IN IP4 88.215.58.7
s=sip call
c=IN IP4 217.163.57.114
t=0 0
m=audio 24194 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

Have you contacted the remote peer, to make they receive your INVITE request, also you sure there is not any firewall blocking outside SIP traffic

Hi, Thanks for your reply. Yeah i have already requested logs from the service provider and asked if they can see the INVITE.

This is just running on my home network at the moment so i’ve only got a basic router with outbound traffic allowed to anywhere. I don’t think it’s my router blocking the traffic because i can accept incoming calls 100% of the time.