Sip 488 (webrtc over sip)

FreePBX Endpoints
1

I am having a devel of time getting web rtc over sip working. So the paste is from asterisk

<— Received SIP request (2796 bytes) from UDP:10.123.245.111:5060 —>
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.123.245.111:5060;branch=z9hG4bKa7258c373cea6e20feaa5d431815dfa2.0;sfent
Session-Expires: 14400
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK666521.gpgDL8xf9MKZcM2jcZVVjOt88sjyc-eWEB0AQDYvnS0_
Max-Forwards: 68
To: sip:*[email protected]
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
Call-ID: s0986vrpbcp3h1d7pirf
CSeq: 9774 INVITE
Contact: sip:edSRgxVhVIdC4YC0dmsmPdP9QB4bni2CCzQmVHyyMzSTIvE0NNUBwZlMkodaVRGw_grt5cMSvcBJce1DlXW3axDVDv[email protected]10.123.245.111
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: ice,replaces,outbound
User-Agent: JsSIP 3.10.0
Content-Length: 1918
Record-Route: sip:[email protected];lr

v=0
o=- 746846767847031410 2 IN IP4 10.123.245.111
s=-
c=IN IP4 10.123.245.111
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 87e66346-be50-4f2a-a083-0537a211230b
m=audio 58266 RTP/AVP 111 63 9 0 8 13 110 126
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:1402670331 cname:zBkMFBPWFwfxFwE0
a=ssrc:1402670331 msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a
m=audio 58266 RTP/SAVP 111 63 9 0 8 13 110 126
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:1402670331 cname:zBkMFBPWFwfxFwE0
a=ssrc:1402670331 msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a

<— Transmitting SIP response (738 bytes) to UDP:10.123.245.111:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.123.245.111:5060;rport=5060;received=10.123.245.111;branch=z9hG4bKa7258c373cea6e20feaa5d431815dfa2.0;sfent
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK666521.gpgDL8xf9MKZcM2jcZVVjOt88sjyc-eWEB0AQDYvnS0_
Record-Route: sip:[email protected]:5060;lr
Call-ID: s0986vrpbcp3h1d7pirf
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
To: sip:*[email protected];tag=z9hG4bKa7258c373cea6e20feaa5d431815dfa2.0
CSeq: 9774 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1684415635/43994a875c524e3c3d9da73153004e37”,opaque=“13a0da905124c8c9”,algorithm=MD5,qop=“auth”
Server: TFPBX-16.0.40(20.2.1)
Content-Length: 0

<— Received SIP request (376 bytes) from UDP:10.123.245.111:5060 —>
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.123.245.111:5060;branch=z9hG4bKa7258c373cea6e20feaa5d431815dfa2.0;sfent
Call-ID: s0986vrpbcp3h1d7pirf
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
To: sip:*[email protected];tag=z9hG4bKa7258c373cea6e20feaa5d431815dfa2.0
CSeq: 9774 ACK
Max-Forwards: 70
Content-Length: 0

<— Received SIP request (3081 bytes) from UDP:10.123.245.111:5060 —>
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.123.245.111:5060;branch=z9hG4bK43605d00757e16d0feaa5d431815dfa2.0;sfent
Session-Expires: 14400
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK5412132.CUTN9YcBMR+k7QOdV2Bf0Hngmw6lknqCoA0H9lMGxrM_
Max-Forwards: 68
To: sip:*[email protected]
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
Call-ID: s0986vrpbcp3h1d7pirf
CSeq: 9775 INVITE
Authorization: Digest algorithm=MD5, username=“32897”, realm=“asterisk”, nonce=“1684415635/43994a875c524e3c3d9da73153004e37”, uri=“sip:*[email protected]”, response=“18f6733f57d7bc06b9909857b9bfe16a”, opaque=“13a0da905124c8c9”, qop=auth, cnonce=“sth8eui3983e”, nc=00000001
Contact: sip:edSRgxVhVIdC4YC0dmsmPdP9QB4bni2CCzQmVHyyMzSTIvE0NNUBwZlMkodaVRGw_grt5cMSvcBJce1DlXW3axDVDv[email protected]10.123.245.111
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: ice,replaces,outbound
User-Agent: JsSIP 3.10.0
Content-Length: 1918
Record-Route: sip:[email protected];lr

v=0
o=- 746846767847031410 2 IN IP4 10.123.245.111
s=-
c=IN IP4 10.123.245.111
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 87e66346-be50-4f2a-a083-0537a211230b
m=audio 58268 RTP/AVP 111 63 9 0 8 13 110 126
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:1402670331 cname:zBkMFBPWFwfxFwE0
a=ssrc:1402670331 msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a
m=audio 58268 RTP/SAVP 111 63 9 0 8 13 110 126
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:1402670331 cname:zBkMFBPWFwfxFwE0
a=ssrc:1402670331 msid:87e66346-be50-4f2a-a083-0537a211230b a36035bd-6e0e-446a-99c6-c2121eb53e4a

<— Transmitting SIP response (541 bytes) to UDP:10.123.245.111:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.123.245.111:5060;rport=5060;received=10.123.245.111;branch=z9hG4bK43605d00757e16d0feaa5d431815dfa2.0;sfent
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK5412132.CUTN9YcBMR+k7QOdV2Bf0Hngmw6lknqCoA0H9lMGxrM_
Record-Route: sip:[email protected]:5060;lr
Call-ID: s0986vrpbcp3h1d7pirf
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
To: sip:*[email protected]
CSeq: 9775 INVITE
Server: TFPBX-16.0.40(20.2.1)
Content-Length: 0

== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
<— Transmitting SIP response (1287 bytes) to UDP:10.123.245.111:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.245.111:5060;rport=5060;received=10.123.245.111;branch=z9hG4bK43605d00757e16d0feaa5d431815dfa2.0;sfent
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK5412132.CUTN9YcBMR+k7QOdV2Bf0Hngmw6lknqCoA0H9lMGxrM_
Record-Route: sip:[email protected]:5060;lr
Call-ID: s0986vrpbcp3h1d7pirf
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
To: sip:*[email protected];tag=3196ad17-6ca8-4372-9c59-ee878a250f2f
CSeq: 9775 INVITE
Server: TFPBX-16.0.40(20.2.1)
Contact: sip:10.123.245.20:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 14400;refresher=uas
Content-Type: application/sdp
Content-Length: 443

v=0
o=- 2871771762 4 IN IP4 10.123.245.20
s=Asterisk
c=IN IP4 10.123.245.20
t=0 0
a=group:BUNDLE 0
m=audio 35538 RTP/AVP 0 8 111 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=ssrc:240232346 cname:3ca0268a-bb22-4081-a201-ab5ec3995b7d
a=mid:0
m=audio 0 RTP/SAVP 111 63 9 0 8 13 110 126

<— Received SIP request (584 bytes) from UDP:10.123.245.111:5060 —>
ACK sip:10.123.245.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.245.111:5060;branch=z9hG4bK247933b8b104284bc738187e305a17f7.0
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK2013933.ldQR8Q-wuoIBmRiNWYZepjf3Q1NcvReuIRG3MOj5ITY_
Max-Forwards: 68
To: sip:*[email protected];tag=3196ad17-6ca8-4372-9c59-ee878a250f2f
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
Call-ID: s0986vrpbcp3h1d7pirf
CSeq: 9775 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.10.0
Content-Length: 0

<— Received SIP request (642 bytes) from UDP:10.123.245.111:5060 —>
BYE sip:10.123.245.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.245.111:5060;branch=z9hG4bK7fc621d91fbcd93ac738187e305a17f7.0;sfent
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK7345304.jEB1NZ8kix7XFqK7iOOY1h26v3wKRiftV0GVe+NR79M_
Max-Forwards: 68
To: sip:*[email protected];tag=3196ad17-6ca8-4372-9c59-ee878a250f2f
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
Call-ID: s0986vrpbcp3h1d7pirf
CSeq: 9776 BYE
Reason: SIP ;cause=488; text=“Not Acceptable Here”
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.10.0
Content-Length: 0

<— Transmitting SIP response (501 bytes) to UDP:10.123.245.111:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.245.111:5060;rport=5060;received=10.123.245.111;branch=z9hG4bK7fc621d91fbcd93ac738187e305a17f7.0;sfent
Via: SIP/2.0/WSS 10.123.245.111;branch=z9hG4bK7345304.jEB1NZ8kix7XFqK7iOOY1h26v3wKRiftV0GVe+NR79M_
Call-ID: s0986vrpbcp3h1d7pirf
From: “Daniel Smith” sip:[email protected];tag=aheqh1jl8l
To: sip:*[email protected];tag=3196ad17-6ca8-4372-9c59-ee878a250f2f
CSeq: 9776 BYE
Server: TFPBX-16.0.40(20.2.1)
Content-Length: 0

== Spawn extension (from-internal, 68, 2) exited non-zero on ‘PJSIP/32897-00000034’
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/32897-00000034’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/32897-00000034’
ec2-tel20CLI> quit

The jssip client is from queuemetrics,
my asterisk host is 10.123.245.20
The SBC is at the 10.123.245.111 address internall.
My web client is getting to the point os trying to start the call up but I’m geting the SIP 488 error which suggests something in the sdp is off.

I’m expecting the audio from teh asterisk server to be in the port 20K to 40K and the port range to be in the 58024 o 60999 from the sbc to the web client. (all udp)

2

What is this SBC? What is it expected to do? The SDP is kind of like WebRTC SDP, but also not.

Asterisk also does not support multiple audio streams. It’ll only do one.

3

its an Ingate Siperator using there 6.4 os.

4

Thanks for spotting the multi channels,
So I got ride of the multi audio channels can call out. but not inbound.
So when I answer the call it immediately goes to voicemail,

ec2-tel20CLI>
ec2-tel20CLI>
<— Received SIP request (886 bytes) from UDP:10.123.244.144:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.123.244.144:5060;branch=z9hG4bK1f045dcd
Max-Forwards: 70
From: sip:[email protected];tag=as6567394c
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.12.0
Date: Thu, 18 May 2023 14:11:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 1732256413 1732256413 IN IP4 10.123.244.144
s=Asterisk PBX 16.12.0
c=IN IP4 10.123.244.144
t=0 0
m=audio 18346 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (344 bytes) to UDP:10.123.244.144:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.123.244.144:5060;rport=5060;received=10.123.244.144;branch=z9hG4bK1f045dcd
Call-ID: [email protected]:5060
From: sip:[email protected];tag=as6567394c
To: sip:[email protected]
CSeq: 102 INVITE
Server: TFPBX-16.0.40(20.2.1)
Content-Length: 0

== Spawn extension (func-apply-sipheaders, s, 14) exited non-zero on ‘PJSIP/32897-00000055’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
<— Transmitting SIP request (1613 bytes) to UDP:10.123.245.111:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.245.20:5060;rport;branch=z9hG4bKPjc03ce77b-ac89-4cab-b67f-f6ba370a6ada
From: “+18166125555” sip:[email protected];tag=74f2e1fc-da16-4032-ad13-4eaf25afcba4
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: e3451926-89a8-496a-b5a3-319f4bd793b7
CSeq: 30227 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: “+18166125555” sip:[email protected]
Max-Forwards: 70
User-Agent: TFPBX-16.0.40(20.2.1)
Content-Type: application/sdp
Content-Length: 819

v=0
o=- 947927882 947927882 IN IP4 10.123.245.20
s=Asterisk
c=IN IP4 10.123.245.20
t=0 0
a=group:BUNDLE audio-0
m=audio 36608 RTP/AVPF 18 0 8 110 107 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:iSmEuj78muUjLMUdi1Snd/g/CiqrpLYWraV8/bru
a=ice-ufrag:0973be172704bbe6144a0293015b85fb
a=ice-pwd:495a10f93251e1b1497c38ac6478d84e
a=candidate:Ha7bf5de 1 UDP 2130706431 10.123.245.222 36608 typ host
a=candidate:Ha7bf514 1 UDP 2130706431 10.123.245.20 36608 typ host
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1586890231 cname:fd29c2a9-837a-498a-beed-8aa0274326f5
a=mid:audio-0

<— Transmitting SIP response (532 bytes) to UDP:10.123.244.144:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.123.244.144:5060;rport=5060;received=10.123.244.144;branch=z9hG4bK1f045dcd
Call-ID: [email protected]:5060
From: sip:[email protected];tag=as6567394c
To: sip:[email protected];tag=3cfec4ba-3784-4572-9904-788cfbccdbd3
CSeq: 102 INVITE
Server: TFPBX-16.0.40(20.2.1)
Contact: sip:10.123.245.20:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0

<— Received SIP response (400 bytes) from UDP:10.123.245.111:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.123.245.20:5060;received=10.123.245.20;rport=5060;branch=z9hG4bKPjc03ce77b-ac89-4cab-b67f-f6ba370a6ada
From: “+18166125555” sip:[email protected];tag=74f2e1fc-da16-4032-ad13-4eaf25afcba4
To: sip:[email protected]
Call-ID: e3451926-89a8-496a-b5a3-319f4bd793b7
CSeq: 30227 INVITE
Server: SIParator/6.4.2
Content-Length: 0

<— Received SIP response (679 bytes) from UDP:10.123.245.111:5060 —>
SIP/2.0 180 Ringing
Record-Route: sip:[email protected];lr
Via: SIP/2.0/UDP 10.123.245.20:5060;rport=5060;branch=z9hG4bKPjc03ce77b-ac89-4cab-b67f-f6ba370a6ada;received=10.123.245.20
To: sip:[email protected];tag=5iqvgcej8j
From: “+18166125555” sip:[email protected];tag=74f2e1fc-da16-4032-ad13-4eaf25afcba4
Call-ID: e3451926-89a8-496a-b5a3-319f4bd793b7
CSeq: 30227 INVITE
Contact: sip:e7I1BPhbs0S7eq4o4VspPYPjyc7_96dGeeSG7A6Fhn8LIvE0NNUBwZlMkodaVRGw_SmMr2YB9Bv6ufV40CLnxW0YR_[email protected]10.123.245.111
Supported: ice,replaces,outbound
Content-Length: 0

<— Transmitting SIP response (532 bytes) to UDP:10.123.244.144:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.123.244.144:5060;rport=5060;received=10.123.244.144;branch=z9hG4bK1f045dcd
Call-ID: [email protected]:5060
From: sip:[email protected];tag=as6567394c
To: sip:[email protected];tag=3cfec4ba-3784-4572-9904-788cfbccdbd3
CSeq: 102 INVITE
Server: TFPBX-16.0.40(20.2.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:10.123.245.20:5060
Content-Length: 0

<— Received SIP response (437 bytes) from UDP:10.123.245.111:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.123.245.20:5060;rport=5060;branch=z9hG4bKPjc03ce77b-ac89-4cab-b67f-f6ba370a6ada;received=10.123.245.20
To: sip:[email protected];tag=5iqvgcej8j
From: “+18166125555” sip:[email protected];tag=74f2e1fc-da16-4032-ad13-4eaf25afcba4
Call-ID: e3451926-89a8-496a-b5a3-319f4bd793b7
CSeq: 30227 INVITE
Supported: ice,replaces,outbound
Content-Length: 0

<— Transmitting SIP request (451 bytes) to UDP:10.123.245.111:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.245.20:5060;rport;branch=z9hG4bKPjc03ce77b-ac89-4cab-b67f-f6ba370a6ada
From: “+18166125555” sip:[email protected];tag=74f2e1fc-da16-4032-ad13-4eaf25afcba4
To: sip:[email protected];tag=5iqvgcej8j
Call-ID: e3451926-89a8-496a-b5a3-319f4bd793b7
CSeq: 30227 ACK
Max-Forwards: 70
User-Agent: TFPBX-16.0.40(20.2.1)
Content-Length: 0

== Spawn extension (app-missedcall-hangup, 32897, 8) exited non-zero on ‘PJSIP/32897-00000055’
== Everyone is busy/congested at this time (1:0/0/1)
<— Transmitting SIP response (895 bytes) to UDP:10.123.244.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.244.144:5060;rport=5060;received=10.123.244.144;branch=z9hG4bK1f045dcd
Call-ID: [email protected]:5060
From: sip:[email protected];tag=as6567394c
To: sip:[email protected];tag=3cfec4ba-3784-4572-9904-788cfbccdbd3
CSeq: 102 INVITE
Server: TFPBX-16.0.40(20.2.1)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: sip:10.123.245.20:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 286

v=0
o=- 1732256413 1732256415 IN IP4 10.123.245.20
s=Asterisk
c=IN IP4 10.123.245.20
t=0 0
m=audio 38962 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Received SIP request (432 bytes) from UDP:10.123.244.144:5060 —>
ACK sip:10.123.245.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.244.144:5060;branch=z9hG4bK527bd481
Max-Forwards: 70
From: sip:[email protected];tag=as6567394c
To: sip:[email protected];tag=3cfec4ba-3784-4572-9904-788cfbccdbd3
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.12.0
Content-Length: 0

5

The remote side sent a 488, why that is I don’t know. The INVITE SDP looks vaguely correct. Since you have an SBC involved I don’t know what is or is not correct in relation to it.

6

Thanks for Looking. At least one issue is identified.