Hey guys, running FreePBX v2.10.1.9 with Asterisk v 10.12.0. I’ve just got done installing a new BrightHouse SIP Trunk here in Central Florida. Outbound works like a champ but I’m having issues getting my Inbound DID to route properly to my extensions.
Goal: To dial 386-xxx-4723 and have it ring a phone with the directory number 4723.
Here is my configuration:
SIP Trunk:
Trunk Name: BrightHouse
Outbound CallerID: 386xxxxxxx
Dialed Number Manipulation Rules
(prepend) + 8 | 411
(prepend) + prefix | 911
(prepend) + 8 | 911
(prepend) + 8 | NXXNXXXXXX
(prepend) + 8 | 1NXXNXXXXXX
Outgoing Settings
Trunk Name: BrightHouse
PEER Details:
type=peer
qualify=300
progressinbound=yes
outboundproxy=72.31.114.162
nat=yes
host=72.31.114.162
dtmfmode=rfc2833
disallow=all
canreinvite=yes
allow=ulaw
Incoming Settings
All Blank
Inbound Route
Description: test2
DID Number 386xxx4723
CallerID Number: 386xxx4723
Set Destination
Extensions <4723> 4723
That the extent of my configuration:
This is the SIP debug that shows it can’t find the number 386xxx4723 and plays the service message: ss-noservice
With my limited knowledge of Asterisk it sounds like a context issue?
<— SIP read from UDP:72.31.114.162:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 72.31.114.162:5060;branch=z9hG4bKvmnvl902r669u1nrmdavvbkr10
To: sip:[email protected]
From: “Cell Phone CA” sip:[email protected];user=phone;tag=snl_0029258532_NSN_CLIENT
Call-ID: NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
CSeq: 1235 INVITE
Contact: sip:[email protected]:5060;maddr=72.31.114.162;transport=udp
Content-Type: application/sdp
Content-Length: 172
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, PRACK
Supported: 100rel
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 1800
Date: Mon, 15 Apr 2013 20:18:59 GMT
Max-Forwards: 62
P-Asserted-Identity: “Cell Phone CA” sip:[email protected];user=phone
Privacy: none
v=0
o=- 561743976 0 IN IP4 72.31.114.162
s=-
c=IN IP4 72.31.114.162
t=0 0
m=audio 26262 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
— (19 headers 9 lines) —
Sending to 72.31.114.162:5060 (NAT)
Using INVITE request as basis request - NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
Found peer ‘BrightHouse’ for ‘310xxxyyyy’ from 72.31.114.162:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/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 72.31.114.162:26262
Looking for 386xxx4723 in from-trunk-sip-BrightHouse (domain 172.19.55.2)
list_route: hop: sip:[email protected]:5060;maddr=72.31.114.162;transport=udp
<— Transmitting (NAT) to 72.31.114.162:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.31.114.162:5060;branch=z9hG4bKvmnvl902r669u1nrmdavvbkr10;received=72.31.114.162;rport=5060
From: “Cell Phone CA” sip:[email protected];user=phone;tag=snl_0029258532_NSN_CLIENT
To: sip:[email protected]
Call-ID: NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
CSeq: 1235 INVITE
Server: FPBX-2.10.1(10.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
– Executing [386xxx4723@from-trunk-sip-BrightHouse:1] Set(“SIP/BrightHouse-00000033”, “GROUP()=OUT_2”) in new stack
– Executing [386xxx4723@from-trunk-sip-BrightHouse:2] Goto(“SIP/BrightHouse-00000033”, “from-trunk,386xxx4723,1”) in new stack
– Goto (from-trunk,386xxx4723,1)
– Executing [386xxx4723@from-trunk:1] Set(“SIP/BrightHouse-00000033”, “__FROM_DID=386xxx4723”) in new stack
– Executing [386xxx4723@from-trunk:2] NoOp(“SIP/BrightHouse-00000033”, “Received an unknown call with DID set to 386xxx4723”) in new stack
– Executing [386xxx4723@from-trunk:3] Goto(“SIP/BrightHouse-00000033”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/BrightHouse-00000033”, “”) in new stack
Audio is at 16754
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 72.31.114.162:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 72.31.114.162:5060;branch=z9hG4bKvmnvl902r669u1nrmdavvbkr10;received=72.31.114.162;rport=5060
From: “Cell Phone CA” sip:[email protected];user=phone;tag=snl_0029258532_NSN_CLIENT
To: sip:[email protected];tag=as0cf2391e
Call-ID: NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
CSeq: 1235 INVITE
Server: FPBX-2.10.1(10.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 1305829573 1305829573 IN IP4 172.19.55.2
s=Asterisk PBX 10.12.0
c=IN IP4 172.19.55.2
t=0 0
m=audio 16754 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:72.31.114.162:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 72.31.114.162:5060;branch=z9hG4bK2csbcbao92rc9ktaibkki9nrc3-8g9j5
To: sip:[email protected];tag=as0cf2391e
From: sip:[email protected];user=phone;otg=6;tag=snl_0029258532_NSN_CLIENT
Call-ID: NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
CSeq: 1235 ACK
Contact: sip:[email protected]:5060;maddr=72.31.114.162;transport=udp
Date: Mon, 15 Apr 2013 20:18:59 GMT
Max-Forwards: 62
Content-Length: 0
<------------->
— (10 headers 0 lines) —
– Executing [s@from-trunk:3] Wait(“SIP/BrightHouse-00000033”, “2”) in new stack
– Executing [s@from-trunk:4] Playback(“SIP/BrightHouse-00000033”, “ss-noservice”) in new stack
– <SIP/BrightHouse-00000033> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-trunk:5] SayAlpha(“SIP/BrightHouse-00000033”, “386xxx4723”) in new stack
– <SIP/BrightHouse-00000033> Playing ‘digits/3.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/8.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/6.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/x.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/x.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/x.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/4.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/7.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/2.ulaw’ (language ‘en’)
– <SIP/BrightHouse-00000033> Playing ‘digits/3.ulaw’ (language ‘en’)
– Executing [s@from-trunk:6] Hangup(“SIP/BrightHouse-00000033”, “”) in new stack
== Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/BrightHouse-00000033’
– Executing [h@from-trunk:1] Macro(“SIP/BrightHouse-00000033”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/BrightHouse-00000033”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/BrightHouse-00000033”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/BrightHouse-00000033”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/BrightHouse-00000033’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/BrightHouse-00000033’
Scheduling destruction of SIP dialog ‘NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:[email protected]:5060;maddr=72.31.114.162;transport=udp for address/port to send to
set_destination: set destination to 72.31.114.162:5060
Reliably Transmitting (NAT) to 72.31.114.162:5060:
BYE sip:[email protected]:5060;maddr=72.31.114.162;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.19.55.2:5060;branch=z9hG4bK5100be29;rport
Max-Forwards: 70
From: sip:[email protected];tag=as0cf2391e
To: “Cell Phone CA” sip:[email protected];user=phone;tag=snl_0029258532_NSN_CLIENT
Call-ID: NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
CSeq: 102 BYE
User-Agent: FPBX-2.10.1(10.12.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<— SIP read from UDP:72.31.114.162:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.55.2:5060;received=71.43.192.30;branch=z9hG4bK5100be29;rport=5060
From: sip:[email protected];tag=as0cf2391e
To: “Cell Phone CA” sip:[email protected];user=phone;tag=snl_0029258532_NSN_CLIENT
Call-ID: NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315
CSeq: 102 BYE
Contact: sip:[email protected]:5060;maddr=72.31.114.162;transport=udp
Date: Mon, 15 Apr 2013 20:19:12 GMT
Content-Length: 0
<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘NSNSIP-4b7280c8-4b7280cc-1-11-1366057139-11176-1366068315’ Method: ACK
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Reliably Transmitting (NAT) to 72.31.114.162:5060:
OPTIONS sip:72.31.114.162 SIP/2.0
Via: SIP/2.0/UDP 172.19.55.2:5060;branch=z9hG4bK6f1bfe41;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as215a210b
To: sip:72.31.114.162
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.12.0)
Date: Mon, 15 Apr 2013 20:19:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:72.31.114.162:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.55.2:5060;received=71.43.192.30;branch=z9hG4bK6f1bfe41;rport=5060
From: sip:[email protected];tag=as215a210b
To: sip:72.31.114.162;tag=snl_0107691798
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: sip:72.31.114.162:5060;maddr=72.31.114.162;transport=udp
Supported: sip-stun
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
localhost*CLI> sip set debug off
SIP Debugging Disabled
Thanks in advance!!!