When trying to connect to a conference room I am getting
SIP/2.0 603 Declined
I don't know why asterisk is declining the request to enter the conference
room. I am including the portion of the dial plan that is being executed
along with a sip debug information. As stated before I don't see the
Openmeetings Transport agent register or do anything with asterisk, (am I
suppose to?). The server just declines to let me enter into the conference
room and I unable to determine why. Can someone help me out with this?
>From extension.conf
exten => _40011,1,GotoIf($[${DB_EXISTS(open504/room/${EXTEN})}]?ok:notavail)
exten => _40011,n(ok),Confbridge(${EXTEN},default_bridge,omsip_user)
exten => _40011,n(notavail),Hangup
>From Sip Debug
<--- SIP read from UDP:x.x.x.x:49952 --->
REGISTER sip:meetings.glorytoyah.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---e7e629091d6d0a9e;rport
Max-Forwards: 70
Contact: <sip:horace@x.x.x.x:49952;transport=UDP;rinstance=aa89e0d2821e9132>
To: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>
From: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>;tag=f275c241
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4757 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 5.5.3 v2.10.15.0
Authorization: Digest
username="horace",realm="asterisk",nonce="59b69e27",uri="sip:meetings.glorytoyah.org:5060
;transport=UDP",response="19416047ef96e57180711cf3c65efaeb",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to x.x.x.x:49952 (no NAT)
Sending to x.x.x.x:49952 (no NAT)
<--- Transmitting (no NAT) to x.x.x.x:49952 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---e7e629091d6d0a9e;received=x.x.x.x;rport=49952
From: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>;tag=f275c241
To: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>;tag=as1ceb5f1f
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4757 REGISTER
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2db3b07b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'iM97Q5-Kv-TL_VmglcgQug..' in 32000 ms
(Method: REGISTER)
<--- SIP read from UDP:x.x.x.x:49952 --->
REGISTER sip:meetings.glorytoyah.org:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---eee8b07fdd98a3c5;rport
Max-Forwards: 70
Contact: <sip:horace@x.x.x.x:49952;transport=UDP;rinstance=aa89e0d2821e9132>
To: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>
From: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>;tag=f275c241
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4758 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 5.5.3 v2.10.15.0
Authorization: Digest
username="horace",realm="asterisk",nonce="2db3b07b",uri="sip:meetings.glorytoyah.org:5060
;transport=UDP",response="a447e48656b54e723a36d28222efcf83",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to x.x.x.x:49952 (no NAT)
<--- Transmitting (no NAT) to x.x.x.x:49952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x:49952;branch=z9hG4bK-524287-1---eee8b07fdd98a3c5;received=x.x.x.x;rport=49952
From: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>;tag=f275c241
To: <sip:hor...@meetings.glorytoyah.org:5060;transport=UDP>;tag=as1ceb5f1f
Call-ID: iM97Q5-Kv-TL_VmglcgQug..
CSeq: 4758 REGISTER
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:horace@x.x.x.x
:49952;transport=UDP;rinstance=aa89e0d2821e9132>;expires=60
Date: Mon, 23 Aug 2021 05:02:51 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'iM97Q5-Kv-TL_VmglcgQug..' in 32000 ms
(Method: REGISTER)
<--- SIP read from UDP:172.58.69.178:45874 --->
INVITE sip:40...@meetings.glorytoyah.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---52f2ea8febffe027;rport
Max-Forwards: 70
Contact: <sip:horacecell@172.58.69.178:45874;transport=UDP>
To: <sip:40...@meetings.glorytoyah.org>
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.12.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 185
v=0
o=Zoiper 1629694971491 1 IN IP4 172.58.69.178
s=Z
c=IN IP4 172.58.69.178
t=0 0
m=audio 59692 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 9 lines) ---
Sending to 172.58.69.178:45874 (NAT)
Sending to 172.58.69.178:45874 (NAT)
Using INVITE request as basis request - wg7pHc0DwIOKu9v8K_sPHQ..
Found peer 'horacecell' for 'horacecell' from 172.58.69.178:45874
<--- Reliably Transmitting (NAT) to 172.58.69.178:45874 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---52f2ea8febffe027;received=172.58.69.178;rport=45874
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
To: <sip:40...@meetings.glorytoyah.org>;tag=as5df773b0
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 1 INVITE
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b43696f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'wg7pHc0DwIOKu9v8K_sPHQ..' in 32000 ms
(Method: INVITE)
<--- SIP read from UDP:172.58.69.178:45874 --->
ACK sip:40...@meetings.glorytoyah.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---52f2ea8febffe027;rport
Max-Forwards: 70
To: <sip:40...@meetings.glorytoyah.org>;tag=as5df773b0
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:172.58.69.178:45874 --->
INVITE sip:40...@meetings.glorytoyah.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---2fd10fb491da8c9f;rport
Max-Forwards: 70
Contact: <sip:horacecell@172.58.69.178:45874;transport=UDP>
To: <sip:40...@meetings.glorytoyah.org>
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.12.3-mod
Authorization: Digest
username="horacecell",realm="asterisk",nonce="3b43696f",uri="
sip:40...@meetings.glorytoyah.org
;transport=UDP",response="fbb13fe641f45e0320242eaecf96a8ab",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 185
v=0
o=Zoiper 1629694971491 1 IN IP4 172.58.69.178
s=Z
c=IN IP4 172.58.69.178
t=0 0
m=audio 59692 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 9 lines) ---
Sending to 172.58.69.178:45874 (NAT)
Using INVITE request as basis request - wg7pHc0DwIOKu9v8K_sPHQ..
Found peer 'horacecell' for 'horacecell' from 172.58.69.178:45874
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Got SDP version 1 and unique parts [Zoiper 1629694971491 IN IP4
172.58.69.178]
Found RTP audio format 0
Found RTP audio format 101
Found RTP audio format 8
Found RTP audio format 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer -
audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined -
(ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f8ae03505e0 -- Strict RTP learning after remote address set to:
172.58.69.178:59692
Peer audio RTP is at port 172.58.69.178:59692
Peer doesn't provide video
Looking for 40011 in rooms-omsip (domain meetings.glorytoyah.org)
sip_route_dump: route/path hop: <sip:horacecell@172.58.69.178:45874
;transport=UDP>
<--- Transmitting (NAT) to 172.58.69.178:45874 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---2fd10fb491da8c9f;received=172.58.69.178;rport=45874
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
To: <sip:40...@meetings.glorytoyah.org>
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:40011@98.174.244.232:5060>
Content-Length: 0
<------------>
-- Executing [40011@rooms-omsip:1] GotoIf("SIP/horacecell-00000023",
"0?ok:notavail") in new stack
-- Goto (rooms-omsip,40011,3)
-- Executing [40011@rooms-omsip:3] Hangup("SIP/horacecell-00000023",
"") in new stack
== Spawn extension (rooms-omsip, 40011, 3) exited non-zero on
'SIP/horacecell-00000023'
Scheduling destruction of SIP dialog 'wg7pHc0DwIOKu9v8K_sPHQ..' in 32000 ms
(Method: INVITE)
<--- Reliably Transmitting (NAT) to 172.58.69.178:45874 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---2fd10fb491da8c9f;received=172.58.69.178;rport=45874
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
To: <sip:40...@meetings.glorytoyah.org>;tag=as0c939f62
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:172.58.69.178:45874 --->
ACK sip:40...@meetings.glorytoyah.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.0.0.2:34804
;branch=z9hG4bK-524287-1---2fd10fb491da8c9f;rport
Max-Forwards: 70
To: <sip:40...@meetings.glorytoyah.org>;tag=as0c939f62
From: <sip:horacec...@meetings.glorytoyah.org;transport=UDP>;tag=6d96d10f
Call-ID: wg7pHc0DwIOKu9v8K_sPHQ..
CSeq: 2 ACK
Content-Length: 0
<------------->
