MOH is not playing on SIP TServer

Voice & Telephony
1

Hi, I am sure this is a really obvious problem, but I am stuck.

I need help with getting MOH to play on my SIP phone!

I am using X-lite simphones to test SIP TServer and can’t get MOH to work. Can someone please help?
Here is the log of me calling from 6000 to 6002 and then trying to place call on hold!

6000 : 172.30.0.115
6002: 172.30.0.128
MOH and SIP Tserver: 172.30.0.222

Why or why is it not working?

Best regards,
Vic

sipcs: 20:15:44.281 dispatchDE: dlg[161]
sipcs: dialog [161:07@016b6300]: << Event 27 << TRN[191]
Response (0): for CRequest@01B9E1F0 RequestHoldCall-CTISIP_Client[404]/5
– RegisterQueue: CRequest@01B9E1F0 RequestHoldCall-CTISIP_Client[404]/5 01B9F090 21073[0,1]
20:15:44.281 — CIFace::Request —

sipcs: 20:15:44.421 Received [368,UDP] 759 bytes from 172.30.0.128:51646 <<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-89
Contact: sip:6002@172.30.0.128:51646;rinstance=3a3dbffae576e057
To: sip:6002@172.30.0.222;tag=3e78fe4b
From: "Ali2"sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 29AF4546-C3FC-4C30-9A91-AF82B0F1CDED-75@172.30.0.222
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 187

v=0
o=- 7 3 IN IP4 172.30.0.128
s=CounterPath X-Lite 3.0
c=IN IP4 172.30.0.128
t=0 0
m=audio 44882 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=recvonly
a=rtpmap:101 telephone-event/8000

sipcs: 20:15:44.421 dispatchDE: dlg[161]
sipcs: dialog [161:07@016b6300]: << Event 33 << TRN[191]
sipcs: 20:15:44.421 Stored This SDP [161]
sipcs: Acknowledgement

sipcs: 20:15:44.421 Sending [368,UDP] 393 bytes to 172.30.0.128:51646 >>>>>
ACK sip:6002@172.30.0.128:51646;rinstance=3a3dbffae576e057 SIP/2.0
From: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
To: sip:6002@172.30.0.222;tag=3e78fe4b
Call-ID: 29AF4546-C3FC-4C30-9A91-AF82B0F1CDED-75@172.30.0.222
CSeq: 2 ACK
Content-Length: 0
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-89

sipcs: 20:15:44.421 dispatchDE: dlg[161]
sipcs: dialog [161:07@016b6300]: << Event 29 << TRN[191]
sipcs: =unmonitor 4[161]
sipcs: =unmonitor 1[161]

sipcs: 20:15:45.046 Sending [368,UDP] 736 bytes to 172.30.0.115:9969 >>>>>
INVITE sip:172.30.0.115:9969 SIP/2.0
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 INVITE
Content-Length: 187
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88
Contact: sip:172.30.0.222:5060
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=- 7 2 IN IP4 172.30.0.128
s=CounterPath X-Lite 3.0
c=IN IP4 172.30.0.128
t=0 0
m=audio 44882 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly

sipcs: 20:15:45.578 Received [368,UDP] 4 bytes from 172.30.0.222:8630 <<<<<

sipcs: 20:15:46.250 Sending [368,UDP] 736 bytes to 172.30.0.115:9969 >>>>>
INVITE sip:172.30.0.115:9969 SIP/2.0
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 INVITE
Content-Length: 187
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88
Contact: sip:172.30.0.222:5060
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

v=0
o=- 7 2 IN IP4 172.30.0.128
s=CounterPath X-Lite 3.0
c=IN IP4 172.30.0.128
t=0 0
m=audio 44882 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly

sipcs: 20:15:46.531 Received [368,UDP] 393 bytes from 172.30.0.115:1050 <<<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 INVITE
User-Agent: RTC/1.2
Content-Length: 0

sipcs: 20:15:46.531 dispatchDE: dlg[160]
sipcs: dialog [160:07@016f35d0]: << Event 32 << TRN[190]

sipcs: 20:15:46.578 Received [368,UDP] 704 bytes from 172.30.0.115:1050 <<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 INVITE
Contact: sip:172.30.0.115:9969
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 248

v=0
o=- 0 0 IN IP4 172.30.0.115
s=session
c=IN IP4 172.30.0.115
b=CT:1000
t=0 0
m=audio 20386 RTP/AVP 0 8 3 101
a=recvonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

sipcs: 20:15:46.578 dispatchDE: dlg[160]
sipcs: dialog [160:07@016f35d0]: << Event 33 << TRN[190]
sipcs: 20:15:46.578 Stored This SDP [160]
sipcs: Acknowledgement

sipcs: 20:15:46.578 Sending [368,UDP] 384 bytes to 172.30.0.115:9969 >>>>>
ACK sip:172.30.0.115:9969 SIP/2.0
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88

sipcs: 20:15:46.578 dispatchDE: dlg[160]
sipcs: dialog [160:07@016f35d0]: << Event 29 << TRN[190]
sipcs: =unmonitor 4[160]
20:15:46.578 +++ CIFace::Event +++
+++ Pre-event +++
Type EventHeld
Devices: <6000/6000> <6002/6002> <-/->
Calls: 7/006e016782820007/7.016B7488/c:2/r:1 0/none
Parties: <d6000/6000-006e016782820007>.01B9F690/l:1/r:1/Established,Active,Origination
<d6002/6002-006e016782820007>.01B9F320/l:2/r:0/Established,Active,Destination
none
Flags: divert=0 hook=1 postCall=0 active=1 moveAll=1 callType=1
— Pre-event —
Matched with CRequest@01B9E1F0 RequestHoldCall-CTISIP_Client[404]/5
SetContext: for party 6000.01B9F690-006e016782820007.016B7488
+++ Held +++
SetHeld: party 6000.01B9F690-006e016782820007.016B7488, cause Null
@20:15:46.5780 [0] 7.2.001.18 distribute_response: message EventHeld
AttributeEventSequenceNumber 0000000000000085
AttributeTimeinuSecs 578000
AttributeTimeinSecs 1165922146 (20:15:46)
AttributeReferenceID 5
AttributeOtherDNRole 2
AttributeOtherDN ‘6002’
AttributeDNIS ‘6002’
AttributeCallUUID ‘A2EA2E7C-664B-4923-BDAB-F9E76E5B31DE’
AttributeConnID 006e016782820007
AttributeCallID 7
AttributeCallType 1
AttributeCallState 0
AttributeThisDNRole 1
AttributeAgentID ‘10001’
AttributeThisDN ‘6000’
@20:15:46.5780 [ISCC] Debug: Translate: ‘’ → ‘’; result 1 ()
20:15:46.578 Int 04544 Interaction message “EventHeld” generated
20:15:46.578 Trc 04542 EventHeld sent to 404 (0008 CTISIP_Client)
20:15:46.578 Trc 04542 EventHeld sent to 384 (0002 StatServer)
– TellHeld
— Held —
FinishRequest CRequest@01B9E1F0 RequestHoldCall-CTISIP_Client[404]/5
IFace stats: q=8 s=4294967288
– complete
+++ Post-event +++
Type EventHeld
Devices: <6000/6000> <6002/6002> <-/->
Calls: 7/006e016782820007/7.016B7488/c:2/r:0 0/none
Parties: <d6000/6000-006e016782820007>.01B9F690/l:1/r:0/Established,Held,Origination
<d6002/6002-006e016782820007>.01B9F320/l:2/r:0/Established,Active,Destination
none
Flags: divert=0 hook=1 postCall=0 active=1 moveAll=1 callType=1
— Post-event —
– deleted: CRequest@01B9E1F0 RequestHoldCall-CTISIP_Client[404]/5
– AckQueues(): CRequest@01B9E1F0 RequestHoldCall-CTISIP_Client[404]/5 01B9F090
20:15:46.578 — CIFace::Event —
sipcs: =unmonitor 1[160]

sipcs: 20:15:46.578 Received [368,UDP] 704 bytes from 172.30.0.115:1050 <<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 INVITE
Contact: sip:172.30.0.115:9969
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 248

v=0
o=- 0 0 IN IP4 172.30.0.115
s=session
c=IN IP4 172.30.0.115
b=CT:1000
t=0 0
m=audio 20386 RTP/AVP 0 8 3 101
a=recvonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

sipcs: 20:15:46.578 Sending [368,UDP] 384 bytes to 172.30.0.115:9969 >>>>>
ACK sip:172.30.0.115:9969 SIP/2.0
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88

sipcs: 20:15:46.578 Received [368,UDP] 704 bytes from 172.30.0.115:1050 <<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 INVITE
Contact: sip:172.30.0.115:9969
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 248

v=0
o=- 0 0 IN IP4 172.30.0.115
s=session
c=IN IP4 172.30.0.115
b=CT:1000
t=0 0
m=audio 20386 RTP/AVP 0 8 3 101
a=recvonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

sipcs: 20:15:46.578 Sending [368,UDP] 384 bytes to 172.30.0.115:9969 >>>>>
ACK sip:172.30.0.115:9969 SIP/2.0
From: sip:6002@172.30.0.222;tag=909F66E0-C255-4669-BFED-93464F7B08A4-86
To: “Ali2” sip:6000@172.30.0.222:5060;tag=6cd9f616a3fb4960a795ca8e464b88e1;epid=0331eed221
Call-ID: 025dbca0b1494c4ca26222962301226f@172.30.0.115
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-88

2

[tr]
[td]
STREAM MANAGER
[call]
call-address=$HOST
call-protocol=sm

[Log]
all=stdout,network,logs/sm
buffering=true
expire=50
segment=10MB
verbose=all

[x-config]
audio-file-format=wav
file-cache-size=10240
log-trace-flags=-ping -rtp/16th -rtp/nte -rtp/dump +rtcp -xconf -cfgserv
max-mixer-delay=60 ms
max-ports=2000
max-record-file-size=0
max-record-silence=0
max-record-time=300 sec
packet-size=g723=1,g729=2,gsm=2
rtcp-inactivity-timeout=30 sec
rtp-ip-precedence=0
rtp-port=8000
rtp-stream-delay=60
sip-annc-codecs=g723,g729,gsm,pcmu,pcma
sip-conf-codecs=gsm,pcmu,pcma
sip-h261-fmtp=
sip-h263-fmtp=
sip-http-codecs=pcmu
sip-http-delay=120 ms
sip-record-all-conf=false
sip-record-base-name=recording/call
sip-record-codec=pcmu
sip-send-info=auto
sip-port=5061
x-type=other
[/td]
[td]
TSERVER
[TServer]
accept-dn-type=+extension +position +acdqueue +routedn +trunk +routequeue
agent-no-answer-action=none
agent-no-answer-overflow=
agent-no-answer-timeout=15
agent-strict-id=false
am-detected=drop
audio-codecs=telephone-event,PCMU,PCMA,G723,G729,GSM
background-processing=false
background-timeout=60 msec
busy-tone=music/busy_5sec
call-rq-gap=0
cancel-monitor-on-disconnect=true
check-tenant-profile=false
collect-tone=music/collect
compatibility-port=0
consult-user-data=separate
customer-id=
cwk-in-idle-force-ready=true
default-dn=
default-dn-type=none
default-monitor-scope=call
default-music=music/on_hold
dn-del-mode=idle
dtmf-payload=101
external-registrar=
extn-no-answer-overflow=
extn-no-answer-timeout=15
fast-busy-tone=music/atb_5sec
fax-detected=drop
find-trunk-by-location=false
inbound-bsns-calls=false
inherit-bsns-type=false
internal-bsns-calls=false
internal-registrar-domains=
internal-registrar-enabled=true
internal-registrar-persistent=false
intrusion-enabled=true
legal-guard-time=0
logout-on-disconnect=true
log-trace-flags=+iscc +cfg$dn -cfgserv +passwd +udata -devlink -sw -req -callops -conn -client
make-call-alert-info=
management-port=0
max-legs-per-sm=0
max-pred-req-delay=3
merged-user-data=main-only
monitor-internal-calls=true
music-in-queue-file=
mwi-agent-enable=false
mwi-domain=
mwi-extension-enable=false
mwi-group-enable=false
mwi-host=
mwi-port=5061
nas-private=false
notrdy-bsns-cl-force-rdy=false
outbound-bsns-calls=false
override-to-on-divert=false
posn-no-answer-overflow=
posn-no-answer-timeout=15
prd-dist-call-ans-time=
predictive-call-router-timeout=20
recall-no-answer-timeout=15
ringing-on-route-point=true
ring-tone=music/ring_back
router-timeout=10
rq-expire-tmout=32000
session-refresh-interval=1800
silence-tone=music/silence
sip-address=
sip-enable-100rel=true
sip-enable-moh=true
sip-enable-sdp-application-filter=false
sip-enable-sdp-codec-filter=false
sip-enable-sdp-encryption-removal=false
sip-enforce-sdp-origin-rules=false
sip-port=5060
sip-sync-local-contact=
sip-sync-peer-contact=
sm-port=6666
timed-cwk-in-idle=true
unknown-bsns-calls=false
unknown-xfer-merge-udata=false
user-data-limit=16000
wrap-up-time=0
sip-treatments-enabled=true
sip-hold-rfc3264=true

2 Likes
3

Ok, there were several problems with the previous setup -

first of of all, I needed to setup MOH_601 with service-type = treatment.

I did not do it, so it would not even try to use SM.

Of course, now, I am stuck with 404 Error in SM: There is an INVITE when I issue RequestHOLD, but SM logs show that SM returns 404 - not found.

I think the reasons are:

  1. unspecified codec
  2. cannot find file

Application name: StreamManager
Application type: VoIPStreamManager (78)
Command line: sm.exe -service VoIPSM -app StreamManager -host alegria_test -port 2020 -sstart
Host name: alegria_test
DST: TZ = 0, timeb = 0
Time zone: -32400, “Œ‹ž (•W€Žž), “Œ‹ž (•W€Žž)
UTC time: 2006-12-12T12:04:04.218
Local time: 2006-12-12T21:04:04.218
Start time (UTC): 2006-12-12T12:04:04
Running time: 0:00:00:00
Host info: Windows 5.2.3790, 2, Service Pack 1, 1.0, 0110, 3
File: (1) logs/sm.20061212_210404_204.log

21:04:04.218 Trc 04112 The Log Output of type ‘logs/sm’ has been created and opened

21:04:04.218 Std 04503 Connected to ConfigServer at alegria_test:2020 (appName=StreamManager)
@21:04:04.2180 [TCONF] Debug: Main Tenant received: DBID [101], name [HappyTenant]
21:04:04.218 Std 05060 StreamManager started
@21:04:04.2180 [TCONF] Debug: Registering CfgSwitch tenant=101
@21:04:04.2180 [TCONF] Debug: Registering CfgDN tenant=101
@21:04:04.2180 [TCONF] Debug: Registering CfgApplication tenant=101
@21:04:04.2180 [TCONF] Debug: Registering CfgHost tenant=0
@21:04:04.2180 [TCONF] Debug: Registering CfgTenant tenant=0

Stream Manager 7.2.002.02 Compiled: May 31 2006 01:17:27
Genesys Telecommunications Laboratories, Inc., Copyright 1991 - 2006

Build with: TCONF 7.2.005.03
Genesys CommonLib 7.2.000.03
confserv library 7.2.000.03
gservice library 7.2.000.03
Genesys SIP lib = 7.2.000.12

x-config list
options list <+>
@location = NULL
rtp-port = 8000
max-ports = 2000
call list
call-address = ‘$HOST’
call-protocol = sm
rtcp-inactivity-timeout = ‘30 sec’
max-record-time = ‘5 min’ <+>
max-record-silence = ‘0’
max-record-file-size = 0
rtp-ip-precedence = 0
rtp-stream-delay = 60
log-trace-flags = 0x1000022 = -ping -rtp/16th -rtp/nte -rtp/dump +rtcp +xconf +cfgserv <+>
x-type = sm
audio-file-format = wav
file-cache-size = 10240
max-mixer-delay = ‘0.060 sec’ <+>
sip-port = 5061 <+>
sip-annc-codecs = ‘g723,g729,gsm,pcmu,pcma’
sip-conf-codecs = ‘gsm,pcmu,pcma’
sip-http-codecs = ‘pcmu’
sip-http-delay = ‘0.120 sec’ <+>
sip-h261-fmtp = NULL
sip-h263-fmtp = NULL
sip-record-all-conf = false
sip-record-base-name = ‘recording/call’
sip-record-codec = ‘pcmu’
sip-send-info = auto
packet-size = ‘g723=1,g729=2,gsm=2’
remote-server-%d array
DN array

StreamManager work mode: regular
RTP-stream-delay set to 60 msec
RTP-regular-timeout set to 30000 (30 sec)
RTP-record-timeout set to 30000 (30 sec)
RTP port range 8000..9999

Connecting to LCA at port 4999…
Open LCALayer on port=4999
LCA Library version 7.1.100.00

gsip:STACKSIMPLE:INIT:SIP Listener TCP on port 5061 …
(conn_adjust_rlimit) set_sbh_threshold(1016):1
CGCL2Listener[232]: Port 5061 opened for listening, protocol TCP
gsip:STACKSIMPLE:INIT:SIP Listener UDP on port 5061 …
CGCL2Listener[244]: Port 5061 opened for listening, protocol UDP
gsip:STACKSIMPLE:INIT:BASE …
gsip:STACKBASE:INIT:Transaction Manager …
gsip:STACKBASE:INIT:Message Factory …
gsip:STACKBASE:INIT:OK
SIP Stack initialized (port=5061)
sip-annc-codecs = (“g723”,“g729”,“gsm”,“pcmu”,“pcma”)
sip-conf-codecs = (“gsm”,“pcmu”,“pcma”)
sip-http-codecs = (“pcmu”)
RTPLeg[3141593] created RTP:8000(fd=252), RTCP:8001(fd=260)
21:04:04.234 Trc 04541 Registered received from 176 (CfgServer)
21:04:04.234 Trc 04541 Registered received from 176 (CfgServer)
21:04:04.234 Trc 04541 Registered received from 176 (CfgServer)
21:04:04.234 Trc 04541 Registered received from 176 (CfgServer)
21:04:04.234 Trc 04541 Registered received from 176 (CfgServer)

LCALayer: REventRegistered on LCA…

Testing scheduler granularity:
pgtime=404864264(+0) delta=2
pgtime=404864266(+0) delta=2
pgtime=404864268(+0) delta=2
pgtime=404864270(+0) delta=2
pgtime=404864272(+0) delta=2
pgtime=404864274(+0) delta=2
pgtime=404864276(+0) delta=2
RTPtimeWheel tick = 2 msec (60 slots) clock = 404864276

RTPleg[3141593]:8000/8001 completed (remote 0.0.0.0:0)
RX=0/0+0(err=0) rtcp=0(err=0) ssrc[0] jitter=0(max=0)
TX=0/0+0(err=0) rtcp=0(err=0)
21:04:04.281 Std 05061 Initialization completed

GKconnDMX(DMX_Dualmode)::configure(alegria_test:0)

@21:04:04.2810 [TCONF] Debug: Remote Server found: remote-server-%d array <+>
DMX_Dualmode [115] list {aux} <+>
app-type = DMX <+>
hostname = ‘alegria_test’ <+>
TServer list
x-config list

@21:04:04.2810 [TCONF] Debug: Configuration is fully initialized
21:05:09.156 Trc 04541 Message ObjectInfoChanged received from 176 (CfgServer ‘’)
Object dump => CfgDeltaApplication={deltaApplication={DBID=117,name=NIL,password=NIL,type=0,version=NIL,appServerDBIDs=NIL,tenantDBIDs=NIL,isServer=0,serverInfo={hostDBID=0,port=NIL,backupServerDBID=0,timeout=0,attempts=0},options=NIL,state=0,userProperties=NIL,appPrototypeDBID=0,flexibleProperties=NIL,workDirectory=NIL,commandLine=NIL,autoRestart=0,startupTimeout=0,shutdownTimeout=0,redundancyType=0,isPrimary=0,startupType=0,commandLineArguments=NIL},deletedAppServerDBIDs=NIL,deletedTenantDBIDs=NIL,deletedOptions=NIL,changedOptions=(x-config=(audio-file-format=‘.wav’)),deletedUserProperties=NIL,changedUserProperties=NIL,deletedFlexibleProperties=NIL,changedFlexibleProperties=NIL,changedAppServerDBIDs=NIL}
@21:05:09.1710 [TCONF] Debug: Main application changed: DBID [117], name [StreamManager]
21:05:09.187 Std 04106 Log Messages file ‘C:\Program Files\GCTI\SM\sm.lms’ successfully loaded
Network output disabled, cluster undefined

@21:05:09.1870 [TCONF] Error: Configuration option x-config/options/audio-file-format has wrong value ‘.wav’

gsip:CL2LIST[244,UDP]:21:06:11.609 <<<< 484 bytes from 172.30.0.222:5060 <<<<
INVITE sip:MOH_600@172.30.0.222:5061 SIP/2.0
From: "ALEGRIA-6002"sip:6002@172.30.0.222;tag=3b64843d
To: sip:MOH_600@172.30.0.222:5060
Call-ID: 29AF4546-C3FC-4C30-9A91-AF82B0F1CDED-508@172.30.0.222
CSeq: 1 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-559
Contact: sip:172.30.0.222:5060
Alert-Info: Ring Answer
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer

gsip:DLG[1]: INVITE TD = TRN[1]
gsip:STACKBASE:Transport allocated for 172.30.0.222:5060
CGCL2Connector[260]: Connection with 172.30.0.222(172.30.0.222) established
SM_PhoneDialog[1] event 12 INVITE

21:06:11.6090 ---- GKconnSIP[1x]::IncomingDLG(MOH_600) rejected 404

gsip:CL2CONN[260,UDP]:21:06:11.609 >>>> 375 bytes to 172.30.0.222:5060 >>>>
SIP/2.0 404 Not Found

From: "ALEGRIA-6002"sip:6002@172.30.0.222;tag=3b64843d
To: sip:MOH_600@172.30.0.222:5060;tag=8AE7C353-A5FF-4D19-A20D-67049EE8064C-1
Call-ID: 29AF4546-C3FC-4C30-9A91-AF82B0F1CDED-508@172.30.0.222
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bKF7ED18EE-BA12-4D78-8D50-DAF6B93AF672-559;received=172.30.0.222
Content-Length: 0

SM_PhoneDialog[1] event 15 CALLED/ResREJECT
SM_PhoneDialog[1] event 58 DESTROY

4

YES!!! I figured it out!!! (Thanks to Keith!)

Guess what - there is a REALLY important option that Genesys simply forgot to include in 7.2 manual or template.
After adding it, I was able to get the whole thing working!
Thank you for everyone’s help!

5

Vic,

what exactly does sip-enforce-sdp-origin-rules do?

-Koki