;exten = _.,n,Gosub(macro-dumpvars,s,1()) ; in case you ever want to look at all of the Asterisk variables on the CLI
;exten = _.,n,Goto(default,6001,1) ; routes incoming call to extension 6001
;exten = _.,n,Goto(ringroups-custom-1,s,1) ; routes incoming call to Ring Group #1
;exten = _.,n,Goto(voicemenu-custom-2,s,1) ; routes incoming call to Nerd Vittles Demo IVR
exten = _.,n,Goto(voicemenu-custom-1,s,1) ; routes incoming call to Stealth AutoAttendant and then to Ring Group #1
On the Anveo end, I think you have to route the DID to your server's IP address.
I'm guessing that the problem has to do with Anveo Direct trunks not being registered which means the incoming calls are treated as anonymous SIP calls and are being blocked. That explains the busy. Somehow we need to figure a way to add the DIDs (1614number and +1614number) to Asterisk so that it treats them as legitimate calls. billsimon is one of several resident SIP experts so perhaps one of them will take a look and tell us where to go from here.
[anveo_direct_in_67.212.84.21]
host=67.212.84.21
type=peer
insecure=port,invite
context=DID_AnveoDirect
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
[anveo_direct_in_176.9.39.206]
host=176.9.39.206
type=peer
insecure=port,invite
context=DID_AnveoDirect
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
[anveo_direct_in_50.22.102.242]
host=50.22.102.242
type=peer
insecure=port,invite
context=DID_AnveoDirect
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
[anveo_direct_in_50.22.101.14]
host=50.22.101.14
type=peer
insecure=port,invite
context=DID_AnveoDirect
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
[anveo_direct_in_72.9.149.25]
host=72.9.149.25
type=peer
insecure=port,invite
context=DID_AnveoDirect
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
So jrglass... to translate the briankelly63 FreePBX solution into Incredible PBX for Asterisk-GUI syntax, you would cut-and-paste the code below into the bottom of /etc/asterisk/extensions.conf and then reload your dialplan with asterisk-reload. Adjust the call destination by uncommenting one of the options in [DID_AnveoDirect_default] context. By default, incoming calls will ring on Ring Group #1 which is extensions 6001 and 6002. Let us know if that solves it. I don't have any Anveo Direct DIDs at the moment to test this. We'll add this to the base install once it's confirmed to work.
Code:[anveo_direct_in_67.212.84.21] host=67.212.84.21 type=friend insecure=port,invite context=DID_AnveoDirect canreinvite=yes qualify=no disallow=all allow=ulaw [anveo_direct_in_176.9.39.206] host=176.9.39.206 type=friend insecure=port,invite context=DID_AnveoDirect canreinvite=yes qualify=no disallow=all allow=ulaw [anveo_direct_in_50.22.102.242] host=50.22.102.242 type=friend insecure=port,invite context=DID_AnveoDirect canreinvite=yes qualify=no disallow=all allow=ulaw [anveo_direct_in_50.22.101.14] host=50.22.101.14 type=friend insecure=port,invite context=DID_AnveoDirect canreinvite=yes qualify=no disallow=all allow=ulaw [anveo_direct_in_72.9.149.25] host=72.9.149.25 type=friend insecure=port,invite context=DID_AnveoDirect canreinvite=yes qualify=no disallow=all allow=ulaw
When all is said and done we want to be looking at the CLI or logging info to see if anything is actually making it to our switch. Assuming we have the IP's and ports covered from every angle then look at the Anveo SIP trace that is produced in the Anveo CDR list for every call.
I have had a couple of occasions on a new DID where everything was set correctly but the calls were not coming to me. Anveo had to make an adjustment on their end.
$[E164][email protected]
/*>>>|55.110.31.111:5060 @ 2015-01-31 21:42:54 */
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bKb3b38cc2a632cdf9420288bf00a3bb30;rport
Max-Forwards: 70
From: "ABC NY" <sip:[email protected]>;tag=87793541fd7ec56437d39de4e0efa9fa
To: <sip:[email protected]>
Call-ID: [email protected]_1
CSeq: 200 INVITE
Contact: Anonymous <sip:[email protected]:5060>
Expires: 300
User-Agent: Anveo Callcontrol
cisco-GUID: 786934192-3810394587-1589482528-4008995038
h323-conf-id: 786934192-3810394587-1589482528-4008995038
P-Asserted-Identity: <sip:[email protected]:5060>
Diversion: <sip:[email protected]:5060>;privacy=off;screen=no; reason=unconditional; counter=1
X-anveo-e164: 18454772222
Content-Type: application/sdp
Content-Length: 288
v=0
o=Sonus_UAC 687941971 1219615303 IN IP4 67.231.5.112
s=SIP Media Capabilities
c=IN IP4 67.231.5.79
t=0 0
m=audio 46728 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
/*<<<|55.110.31.111:5060 @ 2015-01-31 21:42:54 */
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bKb3b38cc2a632cdf9420288bf00a3bb30;received=50.22.101.14;rport=5060
From: "ABC NY" <sip:[email protected]>;tag=87793541fd7ec56437d39de4e0efa9fa
To: <sip:[email protected]>
Call-ID: [email protected]_1
CSeq: 200 INVITE
User-Agent: FPBX-
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
/*<<<|55.110.31.111:5060 @ 2015-01-31 21:42:54 */
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bKb3b38cc2a632cdf9420288bf00a3bb30;received=50.22.101.14;rport=5060
From: "ABC NY" <sip:[email protected]>;tag=87793541fd7ec56437d39de4e0efa9fa
To: <sip:[email protected]>;tag=as651d8eb0
Call-ID: [email protected]_1
CSeq: 200 INVITE
User-Agent: FPBX-
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
/*<<<|55.110.31.111:5060 @ 2015-01-31 21:42:55 */
SIP/2.0 200 OK
Via: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bKb3b38cc2a632cdf9420288bf00a3bb30;received=50.22.101.14;rport=5060
From: "ABC NY" <sip:[email protected]>;tag=87793541fd7ec56437d39de4e0efa9fa
To: <sip:[email protected]>;tag=as651d8eb0
Call-ID: [email protected]_1
CSeq: 200 INVITE
User-Agent: FPBX-
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 11592 11592 IN IP4 55.110.31.111
s=session
c=IN IP4 55.110.31.111
t=0 0
m=audio 13942 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek: ff - - - -
a=ptime:20
a=sendrecv
/*>>>|55.110.31.111:5060 @ 2015-01-31 21:42:55 */
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 50.22.101.14:5060;rport;branch=z9hG4bK902daaeedfea8310b11f4a63723e7895
Max-Forwards: 70
From: "ABC NY" <sip:[email protected]>;tag=87793541fd7ec56437d39de4e0efa9fa
To: <sip:[email protected]>;tag=as651d8eb0
Call-ID: [email protected]_1
CSeq: 200 ACK
User-Agent: Anveo Callcontrol
Content-Length: 0
/*<<<|55.110.31.111:5060 @ 2015-01-31 21:42:56 */
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 55.110.31.111:5060;branch=z9hG4bK57a33644;rport
From: <sip:[email protected]>;tag=as651d8eb0
To: "ABC NY" <sip:[email protected]>;tag=87793541fd7ec56437d39de4e0efa9fa
Call-ID: [email protected]_1
CSeq: 102 BYE
User-Agent: FPBX-
Max-Forwards: 70
Content-Length: 0
/*>>>|55.110.31.111:5060 @ 2015-01-31 21:42:56 */
SIP/2.0 200 OK
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