Calls coming in from Nexvortex provider but failing

I am receiving calls on my providers DID but they are failing. I see them come in on the CLI.
“Number you have dialed is not in service.” Outgoing calls are fine.
Can you tell why they are failing?

— SIP read from UDP:66.23.190.100:5060 —
INVITE sip:[email protected] SIP/2.0
Record-Route: sip:66.23.190.100;lr
Record-Route: sip:209.193.79.10;lr
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK9e31.ebf0a7cfe873d85e1a5f5d65a3a7784c.0
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK9e31.796d7bc84f67efdc28a2b107092746af.0
Via: SIP/2.0/UDP 209.193.79.30;branch=z9hG4bK9e31.69d7b720ef0a7a7a24c2eff9bdc7a711.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe02914c199680688
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 161423 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Accept: application/sdp
Contact: “ALAMEDA CA” sip:[email protected]:5060
Remote-Party-ID: “ALAMEDA CA” sip:[email protected]:5060;privacy=off
Supported: replaces
Content-Length: 280
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 890987 882100 IN IP4 67.231.5.112
s=SIP Media Capabilities
c=IN IP4 67.231.5.80
t=0 0
m=audio 5804 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 0-15
a=sendrecv
a=ptime:20

— (20 headers 13 lines) —
Sending to 66.23.190.100:5060 (NAT)
Sending to 66.23.190.100:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘nexVortex2’ for ‘15102148501’ from 66.23.190.100:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 67.231.5.80:5804
Looking for 15108981436 in from-trunk (domain xx.xxx.xxx.xx)
sip_route_dump: route/path hop: sip:66.23.190.100;lr
sip_route_dump: route/path hop: sip:209.193.79.10;lr

— Transmitting (NAT) to 66.23.190.100:5060 —
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK9e31.ebf0a7cfe873d85e1a5f5d65a3a7784c.0;received=66.23.190.100;rport=5060
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK9e31.796d7bc84f67efdc28a2b107092746af.0
Via: SIP/2.0/UDP 209.193.79.30;branch=z9hG4bK9e31.69d7b720ef0a7a7a24c2eff9bdc7a711.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe02914c199680688
Record-Route: sip:66.23.190.100;lr
Record-Route: sip:209.193.79.10;lr
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 161423 INVITE
Server: FPBX-13.0.190.15(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0


-- Executing [[email protected]:1] Set("SIP/nexVortex2-000007bb", "__FROM_DID=15108981436") in new stack
-- Executing [[email protected]:2] NoOp("SIP/nexVortex2-000007bb", "Received an unknown call with DID set to 15108981436") in new stack
-- Executing [[email protected]:3] Goto("SIP/nexVortex2-000007bb", "s,a2") in new stack
-- Goto (from-trunk,s,2)
-- Executing [[email protected]:2] Answer("SIP/nexVortex2-000007bb", "") in new stack

Audio is at 14866
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

— Reliably Transmitting (NAT) to 66.23.190.100:5060 —
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK9e31.ebf0a7cfe873d85e1a5f5d65a3a7784c.0;received=66.23.190.100;rport=5060
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK9e31.796d7bc84f67efdc28a2b107092746af.0
Via: SIP/2.0/UDP 209.193.79.30;branch=z9hG4bK9e31.69d7b720ef0a7a7a24c2eff9bdc7a711.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe02914c199680688
Record-Route: sip:66.23.190.100;lr
Record-Route: sip:209.193.79.10;lr
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected];tag=as1aec6f11
Call-ID: [email protected]
CSeq: 161423 INVITE
Server: FPBX-13.0.190.15(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 1373791188 1373791188 IN IP4 xx.xxx.xxx.xx
s=Asterisk PBX 13.13.1
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 14866 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


Retransmitting #1 (NAT) to 66.23.190.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK9e31.ebf0a7cfe873d85e1a5f5d65a3a7784c.0;received=66.23.190.100;rport=5060
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK9e31.796d7bc84f67efdc28a2b107092746af.0
Via: SIP/2.0/UDP 209.193.79.30;branch=z9hG4bK9e31.69d7b720ef0a7a7a24c2eff9bdc7a711.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe02914c199680688
Record-Route: sip:66.23.190.100;lr
Record-Route: sip:209.193.79.10;lr
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected];tag=as1aec6f11
Call-ID: [email protected]
CSeq: 161423 INVITE
Server: FPBX-13.0.190.15(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 1373791188 1373791188 IN IP4 xx.xxx.xxx.xx
s=Asterisk PBX 13.13.1
c=IN IP4 xx.xxx.xxx.xx
t=0 0
m=audio 14866 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


— SIP read from UDP:66.23.190.100:5060 —
ACK sip:[email protected]:5060 SIP/2.0
Record-Route: sip:66.23.190.100;lr
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK9e31.a974d3277a2789e237793d350b6cd3ff.0
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK9e31.2941d136c535076e6d04bdc1150c5468.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe02bdf3c4b603b3f
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected];tag=as1aec6f11
Call-ID: [email protected]
CSeq: 161423 ACK
Max-Forwards: 68
Content-Length: 0


— (11 headers 0 lines) —
– SIP/Vitelity-Outbound-000007ba answered SIP/4000-000007b9
– Channel SIP/Vitelity-Outbound-000007ba joined ‘simple_bridge’ basic-bridge 97303173-4037-4026-bec5-32f1b80f40aa
– Channel SIP/4000-000007b9 joined ‘simple_bridge’ basic-bridge 97303173-4037-4026-bec5-32f1b80f40aa

— SIP read from UDP:66.23.190.100:5060 —
ACK sip:[email protected]:5060 SIP/2.0
Record-Route: sip:66.23.190.100;lr
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK9e31.f41843676ac75f4a8b901d8bd41a1c44.0
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK9e31.8702e46ca5d9fef661608ecfec4768e1.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe02c80f34b603b3f
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected];tag=as1aec6f11
Call-ID: [email protected]
CSeq: 161423 ACK
Max-Forwards: 68
Content-Length: 0


— (11 headers 0 lines) —
0x7fe0d54dbfd0 – Probation passed - setting RTP source address to 10.4.32.184:12254
0x7fdfe4006710 – Probation passed - setting RTP source address to 64.2.142.187:16620
– Executing [[email protected]:3] Log(“SIP/nexVortex2-000007bb”, “WARNING,Friendly Scanner from 66.23.190.100;branch=z9hG4bK9e31.ebf0a7cfe873d85e1a5f5d65a3a7784c.0”) in new stack
[2017-02-12 22:08:46] WARNING[55590][C-00000527]: Ext. s:3 @ from-trunk: Friendly Scanner from 66.23.190.100;branch=z9hG4bK9e31.ebf0a7cfe873d85e1a5f5d65a3a7784c.0
– Executing [[email protected]:4] Wait(“SIP/nexVortex2-000007bb”, “2”) in new stack
0x7fe0d54dea30 – Probation passed - setting RTP source address to 67.231.5.80:5804
Really destroying SIP dialog ‘[email protected]’ Method: BYE
– Executing [[email protected]:5] Playback(“SIP/nexVortex2-000007bb”, “ss-noservice”) in new stack
– SIP/nexVortex2-000007bb Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Channel SIP/4000-000007b9 left ‘simple_bridge’ basic-bridge 97303173-4037-4026-bec5-32f1b80f40aa
– Channel SIP/Vitelity-Outbound-000007ba left ‘simple_bridge’ basic-bridge 97303173-4037-4026-bec5-32f1b80f40aa
== Spawn extension (macro-dialout-trunk, s, 21) exited non-zero on ‘SIP/4000-000007b9’ in macro ‘dialout-trunk’
== Spawn extension (restrictedroute-a1e59edd966174c8f07da9c451f01e1b, 915108981436, 8) exited non-zero on ‘SIP/4000-000007b9’
– Executing [[email protected]:1] Hangup(“SIP/4000-000007b9”, “”) in new stack
== Spawn extension (restrictedroute-a1e59edd966174c8f07da9c451f01e1b, h, 1) exited non-zero on ‘SIP/4000-000007b9’

— SIP read from UDP:66.23.190.100:5060 —
BYE sip:[email protected]:5060 SIP/2.0
Record-Route: sip:66.23.190.100;lr
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK6e31.da5c15db90a0f05a9cc23b33c984ef3e.0
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK6e31.d87dcf073fcdccfed5c4be80e43f1b34.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe03243764b603b3f
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected];tag=as1aec6f11
Call-ID: [email protected]
CSeq: 161424 BYE
Max-Forwards: 68
Content-Length: 0


— (11 headers 0 lines) —
Sending to 66.23.190.100:5060 (NAT)
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: BYE)

— Transmitting (NAT) to 66.23.190.100:5060 —
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.190.100;branch=z9hG4bK6e31.da5c15db90a0f05a9cc23b33c984ef3e.0;received=66.23.190.100;rport=5060
Via: SIP/2.0/UDP 209.193.79.10;branch=z9hG4bK6e31.d87dcf073fcdccfed5c4be80e43f1b34.0
Via: SIP/2.0/UDP 67.231.5.112:5060;branch=z9hG4bK0cBe03243764b603b3f
Record-Route: sip:66.23.190.100;lr
From: “ALAMEDA CA” sip:[email protected];tag=gK0c323458
To: sip:[email protected];tag=as1aec6f11
Call-ID: [email protected]
CSeq: 161424 BYE
Server: FPBX-13.0.190.15(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

A bad provider :slight_smile:

, to which DID is this call intended for? there is no To: to work with.

To: sip:[email protected]

The to header didn’t survive copy paste, I corrected everything in the original post.

Apparently you dont have a matching extension for 15108981436 in your ftom-trunk contest