Hi,
We had a big problem where ASTERISK suddenly seems to have forgotten that Genesys is peered to it. It was rejecting calls because its asking for authentication from Genesys.
Below is the call log:
[Feb 23 08:01:41] VERBOSE[31608] res_pjsip_logger.c: <— Received SIP request (1095 bytes) from UDP:20.72.41.22:5060 —>
INVITE sip:11009054338526@30.74.168.6 SIP/2.0
From: sip:5100@20.72.41.22:5060;tag=00250378-6699-13C9-97BD-2329480AAA77-43746886
To: sip:11009054338526@20.72.41.22:5060
Call-ID: 00250364-6699-13C9-97BD-2329480AAA77-43340259@20.72.41.22
CSeq: 1 INVITE
Content-Length: 204
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.72.41.22:5060;branch=z9hG4bK00250382-6699-13C9-97BD-2329480AAA77-129380191
Contact: sip:5100@20.72.41.22:5060
X-Genesys-CallInfo: routed
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
Max-Forwards: 70
Expires: 60
X-Genesys-GVP-Session-ID: AD565000-85B4-D713-9FF1-66629ADC43EE;gvp.rm.datanodes=1|2;gvp.rm.tenant-id=101_DefaultIVRProfile
X-Genesys-CallUUID: 00HUJNJ6J49SJ5TT4CKKG2LAES0GG7EV
X-ISCC-CofId: location=SIPSwitch_OPS;cofid=20753372
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer
v=0
o=- 1994185493 1 IN IP4 20.73.168.170
s=phone-call
c=IN IP4 10.73.168.170
t=0 0
m=audio 33478 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
[Feb 23 08:01:41] NOTICE[31609] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘sip:5100@20.72.41.22’ failed for ‘20.72.41.22:5060’ (callid: 00250364-6699-13C9-97BD-2329480AAA77-43340259@20.72.41.22) - No matching endpoint found
[Feb 23 08:01:41] VERBOSE[31609] res_pjsip_logger.c: <— Transmitting SIP response (611 bytes) to UDP:20.72.41.22:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.72.41.22:5060;rport=5060;received=20.72.41.22;branch=z9hG4bK00250382-6699-13C9-97BD-2329480AAA77-129380191
Call-ID: 00250364-6699-13C9-97BD-2329480AAA77-43340259@20.72.41.22
From: sip:5100@20.72.41.22;tag=00250378-6699-13C9-97BD-2329480AAA77-43746886
To: sip:11009054338526@20.72.41.22;tag=z9hG4bK00250382-6699-13C9-97BD-2329480AAA77-129380191
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1677110501/1e02135b3a4dcfe1516e14656f62f19f”,opaque=“4009c52f584f6274”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 18.6.0
Content-Length: 0
Error “No matching endpoint found” is observed.
The peer connection between ASTERISK and Genesys is up.
Aor: GenesysOPS01 0
Contact: GenesysOPS01/sip:10.72.41.22:5060 2a2702c068 Avail 2.194
Here is the SIP cnfiguration
[GenesysOPS01]
type=aor
contact=sip:20.72.41.22:5060
qualify_frequency=60
[GenesysOPS01]
transport=transport-udp
type=endpoint
context=TS-ROUTING-OUTBOUND
disallow=all
allow=alaw:30
allow=ulaw:30
aors=GenesysOPS01
sdp_session=TelcoSBC
direct_media=no
[GenesysOPS01]
type=identify
endpoint=GenesysOPS01
match=20.72.41.22/32
Note I changed the IP’s to some random value.
It was previously working until today. We tried reloading pjsip and dialplan and restart ASTERISK many times but issue persisted.