My trunk name is Nextiva, the 9728147186 is the number I’m calling from, and hostname for my trunk is trunking.voipdnsserers.com which resolves to 208.73.146.93
I made the edit to extensions.conf that you suggested, this is what it looks like now…
; from-trunk:
;
; Context is really just an aliax of from-pstn
;
[from-trunk]
include => from-pstn-toheader
Here is the output of the debugging logs when I attempted to call 972-295-9324 and got the same the number you have called is not in service message:
<— SIP read from UDP:208.73.146.93:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK1rf72v305gug5s80m5f1.1
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Call-Id: [email protected]
Contact: sip:[email protected]:5060;transport=udp
Content-Disposition: session; handling=required
Content-Length: 308
Content-Type: application/sdp
CSeq: 1 INVITE
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
In-Reply-To: [email protected]
Supported: timer
To: sip:[email protected]:5060
Max-Forwards: 70
v=0
o=Sonus_UAC 8746 24811 IN IP4 208.73.146.93
s=SIP Media Capabilities
c=IN IP4 208.73.146.93
t=0 0
m=audio 31270 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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=maxptime:20
<------------->
— (14 headers 14 lines) —
Sending to 208.73.146.93:5060 (NAT)
Sending to 208.73.146.93:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘NEXTIVA’ for ‘+19728147186’ from 208.73.146.93:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|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 208.73.146.93:31270
Looking for 861167117 in from-pstn (domain 107.155.85.31)
list_route: hop: sip:[email protected]:5060;transport=udp
<— Transmitting (no NAT) to 208.73.146.93:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK1rf72v305gug5s80m5f1.1;received=208.73.146.93
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.70(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
– Executing [861167117@from-pstn:1] Set(“SIP/NEXTIVA-00000001”, “__FROM_DID=861167117”) in new stack
– Executing [861167117@from-pstn:2] NoOp(“SIP/NEXTIVA-00000001”, “Received an unknown call with DID set to 861167117”) in new stack
– Executing [861167117@from-pstn:3] Goto(“SIP/NEXTIVA-00000001”, “s,a2”) in new stack
– Goto (from-pstn,s,2)
– Executing [s@from-pstn:2] Answer(“SIP/NEXTIVA-00000001”, “”) in new stack
Audio is at 14606
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to 208.73.146.93:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK1rf72v305gug5s80m5f1.1;received=208.73.146.93
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
To: sip:[email protected]:5060;tag=as30ed81f6
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.70(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 992107094 992107094 IN IP4 107.155.85.31
s=Asterisk PBX 11.15.0
c=IN IP4 107.155.85.31
t=0 0
m=audio 14606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<— SIP read from UDP:208.73.146.93:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK8jnu82306gd1sqojp471.1
Call-Id: [email protected]
Content-Length: 0
CSeq: 1 ACK
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
To: sip:[email protected]:5060;tag=as30ed81f6
Max-Forwards: 70
<------------->
— (8 headers 0 lines) —
> 0xb7507050 – Probation passed - setting RTP source address to 208.73.146.93:31270
– Executing [s@from-pstn:3] Log(“SIP/NEXTIVA-00000001”, “WARNING,Friendly Scanner from 208.73.146.93”) in new stack
[2015-07-08 11:19:58] WARNING[23929][C-00000001]: Ext. s:3 @ from-pstn: Friendly Scanner from 208.73.146.93
– Executing [s@from-pstn:4] Wait(“SIP/NEXTIVA-00000001”, “2”) in new stack
– Executing [s@from-pstn:5] Playback(“SIP/NEXTIVA-00000001”, “ss-noservice”) in new stack
– <SIP/NEXTIVA-00000001> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-pstn:6] SayAlpha(“SIP/NEXTIVA-00000001”, “861167117”) in new stack
– <SIP/NEXTIVA-00000001> Playing ‘digits/8.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/6.ulaw’ (language ‘en’)
Reliably Transmitting (NAT) to 71.96.207.234:30910:
OPTIONS sip:[email protected]:30910 SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK20a9c92b;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as135e6ea3
To: sip:[email protected]:30910
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.70(11.15.0)
Date: Wed, 08 Jul 2015 16:20:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<— SIP read from UDP:71.96.207.234:30910 —>
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK20a9c92b;rport=5060
To: sip:[email protected]:30910;tag=325e204d
From: “Unknown” sip:[email protected];tag=as135e6ea3
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
Reliably Transmitting (no NAT) to 208.73.146.93:5060:
OPTIONS sip:trunking.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK549ce56f
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as65e745fb
To: sip:trunking.voipdnsservers.com
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.70(11.15.0)
Date: Wed, 08 Jul 2015 16:20:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<— SIP read from UDP:208.73.146.93:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK549ce56f
From: “Unknown” sip:[email protected];tag=as65e745fb
To: sip:trunking.voipdnsservers.com;tag=aprqngfrt-kq4c0i30000c6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
<------------->
— (6 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/6.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/7.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/7.ulaw’ (language ‘en’)
– Executing [s@from-pstn:7] Hangup(“SIP/NEXTIVA-00000001”, “”) in new stack
== Spawn extension (from-pstn, s, 7) exited non-zero on ‘SIP/NEXTIVA-00000001’
– Executing [h@from-pstn:1] Macro(“SIP/NEXTIVA-00000001”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] ExecIf(“SIP/NEXTIVA-00000001”, “0?Set(CDR(recordingfile)=.)”) in new stack
– Executing [s@macro-hangupcall:2] GotoIf(“SIP/NEXTIVA-00000001”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] Hangup(“SIP/NEXTIVA-00000001”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/NEXTIVA-00000001’ in macro ‘hangupcall’
== Spawn extension (from-pstn, h, 1) exited non-zero on 'SIP/NEXTIVA-00000001’
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 208.73.146.93:5060
Reliably Transmitting (no NAT) to 208.73.146.93:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK78d7d9e0
Max-Forwards: 70
From: sip:[email protected]:5060;tag=as30ed81f6
To: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-12.0.70(11.15.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<— SIP read from UDP:208.73.146.93:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK78d7d9e0
From: sip:[email protected]:5060;tag=as30ed81f6
To: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
Call-ID: [email protected]
CSeq: 102 BYE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘[email protected]’ Method: ACK
pbx*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@pbx ~]#