[SOLVED] PJSIP - No matching endpoint found

Asterisk Asterisk SIP
1

Hi, can anyone help me to find why I’m getting this error when I receive a call on a PJSIP trunk:

[Apr 19 11:24:10] NOTICE[6780]: res_pjsip/pjsip_distributor.c:631 log_failed_request: Request 'INVITE' from '<sip:4734338535@sipgw5016.com;user=phone>' failed for '179.124.44.234:5060' (callid: 6994167190547346602-1524148126-502485748) - No matching endpoint found

Here’s my pjsip.conf section:

[falemaisvoip]
type=registration
outbound_auth=falemaisvoip
server_uri=sip:179.124.44.234
client_uri=sip:2754375@179.124.44.234
auth_rejection_permanent=no

[falemaisvoip]
type=auth
auth_type=userpass
username=2754375
password=notmyrealpassword

[falemaisvoip]
type=aor
contact=sip:179.124.44.234
qualify_frequency=60

[falemaisvoip]
type=endpoint
context=from-pstn
disallow=all
allow=g729,alaw
;auth=falemaisvoip
outbound_auth=falemaisvoip
aors=falemaisvoip
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
from_user=2754375
from_domain=179.124.44.234

And here’s the SIP trace:

<--- Received SIP request (1275 bytes) from UDP:179.124.44.234:5060 --->
INVITE sip:s@131.161.42.186:5060 SIP/2.0
Record-Route: <sip:179.124.44.234;lr;ftag=6994167190547346602;did=2b.c0f91246>
From: <sip:4734338535@sipgw5016.com;user=phone>;tag=6994167190547346602
To: <sip:551130904522@179.124.44.234;user=phone>
Call-ID: 6994167190547346602-1524148126-502485748
CSeq: 1 INVITE
Record-Route: <sip:177.20.193.6;lr>
Via: SIP/2.0/UDP 179.124.44.234:5060;branch=z9hG4bK9e0a.06e4dce1.0
Via: SIP/2.0/UDP 177.20.193.6:5060;rport=5060;received=177.20.193.6;branch=z9hG4bKuq4oaYEUcC-ICmuuq4oaOIKsWC*m!GG8-.1-2c50eba8
Min-SE: 90
Session-Expires: 3600;refresher=uac
Contact: <sip:4734338535@177.20.193.6;user=phone>
Allow: INVITE,CANCEL,BYE,ACK,REFER,INFO
Supported: timer,100rel
Diversion: <sip:0211130904522@sipgw5016.com>;privacy=off;screen=no
Max-Forwards: 68
User-Agent: VCS  5.11.1.0-65
Content-Type: application/sdp
Content-Length: 394

v=0
o=MG4000|2.0 22902 38687 IN IP4 177.20.193.6
s=-
c=IN IP4 177.154.138.3
t=0 0
m=audio 51620 RTP/AVP 18 98 96 97 8 0 101
a=rtpmap:98 G.729a/8000
a=rtpmap:96 G.729ab/8000
a=rtpmap:97 G.729b/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:10
a=X-vrzcap:vbd Ver=1 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR2906 Prot=mgcp App=MG

[Apr 19 11:24:10] NOTICE[6780]: res_pjsip/pjsip_distributor.c:631 log_failed_request: Request 'INVITE' from '<sip:4734338535@sipgw5016.com;user=phone>' failed for '179.124.44.234:5060' (callid: 6994167190547346602-1524148126-502485748) - No matching endpoint found
<--- Transmitting SIP response (804 bytes) to UDP:179.124.44.234:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 179.124.44.234:5060;rport=5060;received=179.124.44.234;branch=z9hG4bK9e0a.06e4dce1.0
Via: SIP/2.0/UDP 177.20.193.6:5060;rport=5060;received=177.20.193.6;branch=z9hG4bKuq4oaYEUcC-ICmuuq4oaOIKsWC*m!GG8-.1-2c50eba8
Record-Route: <sip:179.124.44.234;lr;ftag=6994167190547346602;did=2b.c0f91246>
Record-Route: <sip:177.20.193.6;lr>
Call-ID: 6994167190547346602-1524148126-502485748
From: <sip:4734338535@sipgw5016.com;user=phone>;tag=6994167190547346602
To: <sip:551130904522@179.124.44.234;user=phone>;tag=z9hG4bK9e0a.06e4dce1.0
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1524147850/026bef90536cad71b98116b4ee0e99c2",opaque="447a681521df6d27",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/13.13-cert8
Content-Length:  0


<--- Received SIP request (391 bytes) from UDP:179.124.44.234:5060 --->
ACK sip:s@131.161.42.186:5060 SIP/2.0
Via: SIP/2.0/UDP 179.124.44.234:5060;branch=z9hG4bK9e0a.06e4dce1.0
From: <sip:4734338535@sipgw5016.com;user=phone>;tag=6994167190547346602
Call-ID: 6994167190547346602-1524148126-502485748
To: <sip:551130904522@179.124.44.234;user=phone>;tag=z9hG4bK9e0a.06e4dce1.0
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: SIPTEK Softswitch
Content-Length: 0
1 Like
2

You have no “identify” section that would match on an IP address to know what endpoint to use. The line option[1] could also work but is dependent on the provider properly respecting the SIP RFC. As it is chan_pjsip doesn’t know who the INVITE came from.

[1] http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/

2 Likes
3

Hi @jcolp, it worked, thanks.

4

Hello! Have the same problem.
Where is that “identify” section, can you explain please?

5