Hi All,
I am trying to create an Inbound route destined to a Ring Group through a SIP trunk. I am able to call the extensions, but unable to call a Ring Group or an IVR through the Inbound Route. I am really not sure, what i am missing. When the DID for the IVR or Ring Group is called, getting the message from the Asterisk that “the call cannot be completed, please check your number”.
The Inbound Route configuration for the IVR :-
- DID Number : 2000
- Ring Groups : RG<600>
SIP Peer details :-
host=20.1.1.170
type=friend
port=5060
nat=no
disallow=all
allow=ulaw,alaw
qualify=yes
canreinvite=yes
context=from-trunk
When 2000, is dialled, the DID in the SIP Invite is the same, but still getting the same error message.
SIP Logs :-
Invite to the DID 2000 for Ring Group ---->
100 Trying <-----
183 Session Progess <----- (Playing the error message)
<— SIP read from UDP:20.1.1.170:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c
From: sip:[email protected];tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To: sip:[email protected]
Date: Fri, 11 Sep 2015 14:06:41 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 1376285696-0000065536-0000002594-2852192532
Session-Expires: 1800
P-Asserted-Identity: sip:[email protected]
Remote-Party-ID: sip:[email protected];party=calling;screen=yes;privacy=off
Contact: sip:[email protected]:5060;bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 198
v=0
o=CiscoSystemsCCM-SIP 787014 1 IN IP4 20.1.1.170
s=SIP Call
c=IN IP4 20.1.1.170
t=0 0
m=audio 25986 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (22 headers 9 lines) —
Sending to 20.1.1.170:5060 (no NAT)
Sending to 20.1.1.170:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘2723’ for ‘2723’ from 20.1.1.170:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), 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 20.1.1.170:25986
Looking for 2000 in from-internal (domain 20.1.1.58)
list_route: hop: sip:[email protected]:5060
<— Transmitting (NAT) to 20.1.1.170:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060
From: sip:[email protected];tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-12.0.76(11.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
Audio is at 16598
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 20.1.1.170:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060
From: sip:[email protected];tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To: sip:[email protected];tag=as3e6a1653
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-12.0.76(11.19.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 228
v=0
o=root 881046367 881046367 IN IP4 20.1.1.58
s=Asterisk PBX 11.19.0
c=IN IP4 20.1.1.58
t=0 0
m=audio 16598 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
Anything i am missing here ?
Thanks a lot !