[asterisk-users] Calls to Ring Group not working. FreePBX.

[Agasthian P]

Hi All,

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 directly, but unable to call a
Ring Group or an IVR through the Inbound Route config. 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". I am doing the configuration using FreePBX and
the Asterisk version is 12.

The Inbound Route configuration for the IVR :-

   1. DID Number : 2000
   2. 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 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:2000 at 20.1.1.58:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c
From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To:
Date: Fri, 11 Sep 2015 14:06:41 GMT
Call-ID: 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170
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:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: ;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 -
52087400-5f21dff1-354b2-aa010114 at 20.1.1.170
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:

<--- 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: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To:
Call-ID: 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170
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:
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: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395
To: ;tag=as3e6a1653
Call-ID: 52087400-5f21dff1-354b2-aa010114 at 20.1.1.170
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:
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 ? Also please let me know, if you need any other
logs to help me in this.

Thanks a lot !

Agasthian P
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