I have one extension that gets a sip 488 on every call Not sure why but can call inbound to that
pjsip debug info
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.228:5060;branch=z9hG4bK-524287-1—72ed8fb0ef15d299;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=UDP>
To: <sip:[email protected]>
From: <sip:[email protected];transport=UDP>;tag=6005011c
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.4.6 rv2.10.10.2
Allow-Events: presence, kpml, talk
Content-Length: 227v=0
o=Z 1600112593446 1 IN IP4 10.119.101.228
s=Z
c=IN IP4 10.119.101.228
t=0 0
m=audio 20000 RTP/AVP 18 101 0 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv<— Transmitting SIP response (512 bytes) to UDP:10.119.101.228:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.119.101.228:5060;rport=5060;received=10.119.101.228;branch=z9hG4bK-524287-1—72ed8fb0ef15d299
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
From: <sip:[email protected]>;tag=6005011c
To: <sip:[email protected]>;tag=z9hG4bK-524287-1—72ed8fb0ef15d299
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1600112593/c8ae2fc76df0137b0f927842f56bf0c4”,opaque=“7c4f221a0c4ae2ab”,algorithm=md5,qop=“auth”
Server: FPBX-15.0.16.73(16.12.0)
Content-Length: 0<— Received SIP request (356 bytes) from UDP:10.119.101.228:5060 —>
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.228:5060;branch=z9hG4bK-524287-1—72ed8fb0ef15d299;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=z9hG4bK-524287-1—72ed8fb0ef15d299
From: <sip:[email protected];transport=UDP>;tag=6005011c
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
CSeq: 1 ACK
Content-Length: 0<— Received SIP request (1090 bytes) from UDP:10.119.101.228:5060 —>
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.228:5060;branch=z9hG4bK-524287-1—113634197e59905a;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=UDP>
To: <sip:[email protected]>
From: <sip:[email protected];transport=UDP>;tag=6005011c
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.4.6 rv2.10.10.2
Authorization: Digest username=“32481”,realm=“asterisk”,nonce=“1600112593/c8ae2fc76df0137b0f927842f56bf0c4”,uri=“sip:[email protected];transport=UDP”,response=“e0032f69f1723c85eccadf7bb9834397”,cnonce=“611e5946972284a608bc3728932daf57”,nc=00000001,qop=auth,algorithm=md5,opaque=“7c4f221a0c4ae2ab”
Allow-Events: presence, kpml, talk
Content-Length: 227v=0
o=Z 1600112593446 1 IN IP4 10.119.101.228
s=Z
c=IN IP4 10.119.101.228
t=0 0
m=audio 20000 RTP/AVP 18 101 0 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv == Setting global variable ‘SIPDOMAIN’ to ‘10.123.245.18’
<— Transmitting SIP response (320 bytes) to UDP:10.119.101.228:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.119.101.228:5060;rport=5060;received=10.119.101.228;branch=z9hG4bK-524287-1—113634197e59905a
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
From: <sip:[email protected]>;tag=6005011c
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: FPBX-15.0.16.73(16.12.0)
Content-Length: 0<— Transmitting SIP response (374 bytes) to UDP:10.119.101.228:5060 —>
SIP/2.0 488 Not Acceptable Here
I have 2 extensions that can call out but don’t receive calls. In the console I see the pjsip ringing that extensions. I haven’t’ got a debug captures.
I have one other that on some number of calls, they get a busy signal to there DID> Not all the time but like 40% of the time.
Any help would be great. I’m underwater with conversion to freepbx from bare asterisk.