SIP Trunk not accepting inbound calls

I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX                       -1         1124         yes
101                              20001      45           no
102                              20003      18           no
103                              20005      45           no
104                              20006      45           no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0417XXXXXX
Called Number            : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name         : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port    (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port    (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 21
Disconnect Cause (SIP)   : 403
Any Ideas
Doug

Hi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1

Similar Messages

Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working

Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad

Hi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP     LocalIP                                RemoteIP
1     716        717        19314    4000     192.168.1.10                           192.168.1.241
2     717        716        19234    54932    10.249.13.130                          10.224.42.72
Found 2 active RTP connections

SIP trunk incoming and outgoing calls issue

Hi Everyone,
We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
Our call flow is as
IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
Please if some help in sorting out this issue, Thanks in Advance
Tasneem

Inbound call>>>>
The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Date: Thu, 16 Jan 2014 13:31:34 GMT
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 2507
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
Contact:
Record-Route:
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0
AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 504 Gateway Timeout
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Reason: Q.850;cause=102
Content-Length: 0
I suggest you disable this parameter..and test again
voice service voip
sip
rel1xx disable
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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working

This is the setup. Currently in lab environment for a client, but needs to go into production
IP Phone -> CUCM 9 -> H323 GW -> SIP Trunk -> Proprietary device -> Analog phone
Calls complete both ways with no issues. Proprietary devices only uses G711ulaw, so I have configured a xcoder on the H323 GW to transcode to G729 across the WAN link (between the CUCM cluster and the H323 GW).
Pressing keys/sending DTMF tones from the IP phone are not heard in the analog phone
Running a debug voice ccpai inout at the H323 gateway shows me that the DTMF tones are being received the GW and are being sent along. See below:
Seaport#
Seaport#
Seaport#! Pressing digit "9" on VoIP phone
Seaport#
Seaport#
Seaport#
Seaport#
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit "9" on VoIP phone                " on VoIP phone                 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit "       5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
Seaport#
Seaport#
However, debug ccsip does not give me any indications that the DTMF tone is being sent out the SIP trunk. Debug ccsip all attached.
Relevant portions of the H323 configuration are below
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1                                   <- interface to proprietary device
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/2                                  <-interface to Local LAN supporting IP Phones
ip address 10.10.10.254 255.255.255.0
duplex auto
speed auto
sccp local GigabitEthernet0/2
sccp ccm 10.10.10.254 identifier 1 priority 1 version 3.1
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate profile 10 register xcoder_1
dspfarm profile 10 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer voice 2 voip
          description Default Incoming Dial Peer
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric h245-signal rtp-nte
dial-peer voice 6 voip
destination-pattern 90052..                      <- DN of analog phone
session protocol sipv2
session target ipv4:192.168.200.1            <- IP of proprietary device
codec g711ulaw
no vad
sip-ua
registrar ipv4:172.16.88.254 expires 3600
no transport tcp
telephony-service
sdspfarm units 4
sdspfarm transcode sessions 2
sdspfarm tag 1 xcoder_1
I also ran the debug voip rtp session named-event all but nothing was displayed when I pressed the digits on the IP Phone.
Jeff

Please configure "dtmf-relay rtp-nte" command under SIP dial-peers.
Jorge Armijo
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RightFax VM not taking inbound calls

I know I am not going to provide a whole lot of info since there is so much to post so I thought I would lay out what works and what doesn't in hopes someone can guide me to solution.
We have a current physical RF (RightFax) with a tech line coming straight into it avoiding CUCM.
I have (with help) virtualized the RF and have it going thought the CUCM 8.6 through a 2901 voice router
I can dial out through an existing PRI. I have added a new PRI to accept incoming calls and this is where I am stuck. Internally I can set up a four-digit extension...say 2901...and get to the RF virtual server. I pointed the DID to the new PRI and get the below results but the call rings not allocated:
001078: Jul 8 12:10:21.663 EDT: ISDN Se0/2/0:23 Q931: RX <- SETUP pd = 8 callref = 0x00BA
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Facility i = 0x9F8B0100A10F02010106072A8648CE1500040A0100
vgmducgvg01#
Protocol Profile = Networking Extensions
0xA10F02010106072A8648CE1500040A0100
Component = Invoke component
Invoke Id = 1
Operation = InformationFollowing (calling_name)
Name information in subsequent FACILITY message
Calling Party Number i = 0x0081, 'FULL NUMBER'
Plan:Unknown, Type:Unknown
Called Party Number i = 0xC1, '2901'
Plan:ISDN, Type:Subscriber(local)
001079: Jul 8 12:10:21.667 EDT: ISDN Se0/2/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x80BA
Cause i = 0x8081 - Unallocated/unassigned number
When the DID was pointed (mistakenly by the Telcom) to our local PRI, it actually worked but not to our new PRI. The settings mirror each other on the two PRIs to boot. Same Telcom and everything.
The internal ext works but not the full number.
Anyone have any thoughts on where I should look?
THANKS!!!

Ok - so the call is routed into the network and then when you assign it to a route pattern, which is used to route the call to the RightFax server, the call fails. In that case, you will need to look at the H323 integration between RightFax and CUCM. First, you'll need to make sure that the versions are compatible. There are some gotchas when using H323 between CUCM and RightFax just as there are gotchas if you use SIP between CUCM and RightFax. Personally, I'd try the SIP integration over H.323 but it's not that one is inherently better than the other. You need to make sure that you configure the CUCM side based on the latest RightFax integration guide. You also have to make sure that the RightFax is configured appropriately as well. It needs to 1) be configured to accept calls from the CUCM via either SIP or H323 and 2) have the fax DIDs (or abbreviated extensions - whatever you use) configured so it can process the call as well.

Inter-Trunk not route incoming calls from out

Hi,
I setup one extra gateway where I try to route part of our calls. So far I have success to route internal calls into there, but when I'm making a test call from outside that ends into "number is not used" problem.
I have:
- Route ready, elsewere the internal calls are not working.
- PSTN usage, linked to the Route
- Trunk configuration where I have selected the PSTN usage
- Incoming numbers are coming in E164 format
I have also tested the "Test-CsInterTrunkRouting" and that gives "pass":
FirstMatchingRoute : Description=;NumberPattern=^\+358123654789;Name=Test
                     Gateway;SuppressCallerId=False;AlternateCallerId=
MatchingUsage      : Test PSTN Usage
MatchingRoutes     : {Description=;NumberPattern=^\+358123654789;Name=Test
                     Gateway;SuppressCallerId=False;AlternateCallerId=}
But still, when I made a call from outsited the OCSLogger shows that mediation server try to offer call to Front-End which says only: "SIP/2.0 404 Not Found" and then bye-bye.
What is the missing magic, which made the mediation server to see alternative route? I hope it is not required that mediation server must be collocated on the Front Ends, as that one I do not have.
Any good ideas?
ps.
I'm not sure does it matter, but my Lync gives "SIP/2.0 403 Forbidden" when there is coming call from extra gateway. But as the calls into there works, then I don't see why external calls should not also work.
Petri

Could it be even so, that intra-trunk routing requires consolidated mediation server? As the call is owned by the Mediation server (stanalone), and it is trying to offer that to FE. FE reply "does not exist". Because of the standalone Mediation
server does not have the call routing engine like FE have, the call is lost.
I started to think above as Lync users are able to call to that number. So FE is able to do the routing and get calls into the correct place.
I have to say also, I have read
Ken's blog about inter-trunk routing, I have to say that I'm not so sure what he means by this: "Fortunately, in most cases, adding PSTN usages to the trunk has no effect, since there is almost always a Lync user assigned to the incoming phone
numbers". Why to add additional routing for the numbers which are already inuse? I hope it is not required, that you need to have a users ID for each number you do the inter-trunk routing?
Petri

I'm unable to accept inbound calls on my iPhone 5 for a holden commodore VE - 2012 model - any ideas on how to resolve?

How do I resolve the issue for answering inbound calls on my iPhone 5 when connected to Bluetooth in a holden commodore VE - 2012 model. Outbound calls connect ok.

I'm having the same problem with a 2013 GMC terrain. I don't know if the problem is the car or the iPhone 5. I have been seeing this problem more and more on threads. Can anyone help us with this? Is there any info online from Apple re: new updates that would solve this problem?

SNMP for devices that do not accept inbound snmp

Hi All,
I have a requirement to monitor a network device (Cisco) which doesn't accept SNMP requests (161) but it will send out snmp traps on port 162.
I have setup my SCOM 2012 R2 server (running on windows 2008 r2) as per many of the blogs guide, but I think the reason that I am not getting any alerts through is that you need to discover the network device via the snmp protocol before you can monitor
it (ucmp not enough).
Has anyone had a similar situation and found a solution? Would a SMNP proxy work?
Please see the bottom comment on this blog where someone else has asked the same question but without a real solution.
http://kevingreeneitblog.blogspot.co.uk/2012/01/scom-2012-network-monitoring-magic.html

Terry,
As you figured out already, you do have to be able to discover the device using SNMP get. Otherwise SCOM will not accept traps.
Yes, you could use a SNMP proxy. There are just forwarders or ones that write the trap into the event log from where you could get them using an event log rule/monitor.
Or you could use System Center Orchestrator and raise an alert in SCOM directly.
Cheers,
Patrick
Please remember to click “Mark as Answer” on the post that helped you.
Patrick Seidl (System Center and Private Cloud)
Website: http://www.syliance.com
Blog: http://www.systemcenterrocks.com

SIP trunk INCOMING/OUTGOING calling issue

Dears,
i am facing issue i.e inbound / outbound calls are not wokring after configuring SIP trunk . the call flow is
SERVICEPROVIDER--------sip trunk------VG --------------sip trunk ----------CUCM
For Outbound calls ---error-----tot tot tot
For Inbound Calls -----error--- Silence
Below are the configuration and debugs:

Appreciate if some provide me Sip configuration and troubleshooting guide...Thankyou

Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

Hi to all
i have something strange here and i need your assistance
Call Flow:
Sip trunk-->UC540--> CUE
When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
Incoming voicemail is working fine
Incoming calls to phones also ok
Uc540: 8.6
CUE: 8.6.5
A number: 99999999
B number: 22777777
Voice Mail Number:111
Attached is the trace
i see that we hit the correct dial peers .
I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
voice service voip
ip address trusted list
ipv4 172.16.80.0 255.255.255.0
ipv4 172.16.81.0 255.255.255.0
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
no update-callerid
dial-peer voice 1000 voip
description **SIP TRUNK**
translation-profile incoming SIP-INCOMING
translation-profile outgoing SIP-OUTGOING
destination-pattern 9T
modem passthrough nse codec g711alaw
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711alaw
no vad
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
dtmf-relay sip-notify
codec g711ulaw
no vad
Regards
chrysostomos

Hi
Interface                  IP-Address      OK? Method Status                Protocol
FastEthernet0/0            unassigned      YES NVRAM up                    up
FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
FastEthernet0/0.20         10.151.5.130    YES NVRAM up                    up ------> For sip trunk
In0/0                      10.1.10.2       YES unset up                    up    --------> default gw for cue
Vlan1                      unassigned      YES unset up                    up
Vlan100                    unassigned      YES unset up                    up
Vlan200                    unassigned      YES unset up                    down
Vlan300                    unassigned      YES unset up                    down
NVI0                       10.1.10.2       YES unset up                    up
BVI1                       192.168.20.1    YES NVRAM up                    up
BVI100                     10.1.1.1        YES NVRAM up                    up   ---------> ip for cme
Loopback0                  10.1.10.2       YES NVRAM up                    up   ------> default gw for cue
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
voice-class sip bind control source-interface BVI100
voice-class sip bind media source-interface BVI100
dtmf-relay sip-notify
codec g711ulaw
no vad
interface FastEthernet0/0.10
description **FOR INTERNET**
encapsulation dot1Q 10
ip address dhcp
ip access-group 105 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
interface FastEthernet0/0.20
description **FOR SIP TRUNK WITH ISP**
encapsulation dot1Q 20
ip address 10.151.5.130 255.255.255.240
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
ping 10.1.10.1 source bvi100
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
Packet sent with a source address of 10.1.1.1
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
I have bind the interface of cme ( 10.1.1.1) but the call fails again
Attached is the trace
Anything to advice?

Unable to perform call transfer & call park through SIP Trunk (SKYPE)

The Scenario is:
I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
Anyone has facing the same issue?

MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
HTH
Manish

Accept/not accept in coming call

Hi,
When I get a call sometimes I can choose with a red and a green button between accepting and not accepting the call (I don't know what the english text is on the buttons, in Dutch it is "weigeren" " accepteren")
Sometimes I only get a slider to answer the call.
I want to have the option with the two buttons all the time. How do I do that???
thank you in advance
grtz Roelof-Jan

You only get the red and green buttons on the screen when the phone is unlocked. Otherwise you get a slider. If you wish to reject the call when only the slider appears simply hit the home button.

CTI Problem in inbound call scenario with Siemens OpenScape Contact Center (CMS)

Hello Experts,
we are working on a SAP CRM 7.0 EhP 2 with CTI to a CMS from Siemens (OpenScape Contact Center).
The basic integration works perfect, so that agents get informed about incoming calls in their Interaction Center UI and so on.
Currently we have a problem with handling of incoming calls:
When the agent is available (workmode 'ready') and he does not accept the call, the next available agent in the queue gets the call.
So far, so good. The problem is that the calls, the agent did not accept, do not disappear in the upper right corner. They are counted up to 5 and a timer starts counting up to 600 seconds. When the 600 seconds are reached, the agend gets disconnected from the communication channel, too.
(In case the agent talkes a call again, this counter is reset to again)
Is there any possibility to avoid the signalization of unaccepted calls from CRM side (generally)?
How could we prevent the agent from disconnection after 600 seconds and 5 unaccepted calls from CRM side?
Thanks in advance!
Kind regards
Martin
Message was edited by: Joaquin Fornas

Hi Yoro,
do you click on a physical phone button or on the button in the WebUI.
As far is i know, the OpenScape Contact Center does not support hang up funktionallity of the related button in the WebUI.
If you do not use wrapup time (which is used to give the agent time for the data maintenance before he continues receiving calls) the Counter shoud end after the call.
It seams that the OpenScape does not work propperly.
For more details you can take a look into the Log:
Transaction CRM_ICI_TRACE
You have to set the User Parameter CRM_ICI_TRACELEVEL to value XXX to enable the logging.
Good Luck!
Kind regards,
Martin Gaschk

Why do we need MTP in the SIP trunk for CVP warm transfers

Hi All,
Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
Thanks in advance!!
Regards,
Thammaya Gupta K.

I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
@ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
I have seen a documentation states:
" To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
Our ultimate goal is to use g729 without using HW MTP/ transcoder.
IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP

Can not make srtp call to CISCO-GW

Hi,
There are one CISCO2821 and one CISCO3845. Both of them are configured with SRTP. They can make calls to the third-party phone with SRTP. But both gateways can not accept any call if I set them to use SRTP only. Even if the 2 gateways call each other, the calls still can not work. If I set the gw to allow srtp falling back to rtp , the calls can be established. The attached file is the debugging log from the cisco2821 .
Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 12.4(15)T
6, RELEASE SOFTWARE (fc2)
Cisco IOS Software, 3800 Software (C3845-ADVIPSERVICESK9-M), Version 12.4(15)T5,
RELEASE SOFTWARE (fc4)
Thanks lot of for your help

Upgrading to 12.4(20) T can resolve this problem. This new version adds a new option which the document doesn't point out. It took a lot of time.

SIP Trunk not accepting inbound calls

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