UC560 SIP Configuration via CCA

We are trying to configure SIP trunks VIA CCA (Latest version) on a UC560 latest version.
However the username and password field doesn't save. When you open CCA SIP trunk configuration the fields are blank, fill them in and save, exist and go back in they are still blank. If you ftp the config off the unit and search for the field "username" only the phones and admins come up - no mention of SIP.
Are we missing something?

Hello,
This sounds like you may have Windows firewall or Anti-virus software running. Also make sure you are using Windows 7 or XP. Windows 8 will not work properly with CCA.
Also if you are using SSL VPN, make sure that only version Anyconnect 2.5.6005 is loaded on the UC. Anything higher will use up the memory and you will not be able to write to the config.
I hope this helps.
Regards,
Chris

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Please help with SIP configuration on 2801 router

Hi All.
Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.
The information from our SIP provider:
We have issued the following DDI range: 018877000 – 99
There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.
Configuration details are as follows:
Our Primary Proxy:-        99.234.56.78
Codec supported:-             G711Alaw, G729 (G711Alaw is the preferred codec)
Fax Support:-                     T38 and G711Alaw
DTMF:-                                 RFC2833 and INFO
CLI Method:-                     Remote-Party-ID
Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.
This is a SIP configuration on Cisco2801 router (I used outgoing calls only):
translation-rule 10
Rule 0 ^90 0
Rule 1 ^91 1
Rule 2 ^92 2
Rule 3 ^93 3
Rule 4 ^94 4
Rule 5 ^95 5
Rule 6 ^96 6
Rule 7 ^97 7
Rule 8 ^98 8
Rule 9 ^99 9
interface FastEthernet0/0.1
description ***DATA VLAN***
encapsulation dot1Q 1 native
ip address 10.1.1.101 255.255.255.0
interface FastEthernet0/0.2
description ***VOICE VLAN***
encapsulation dot1Q 2
ip address 192.168.22.1 255.255.255.0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
call start slow
sip
bind control source-interface FastEthernet0/0.2
bind media source-interface FastEthernet0/0.2
registrar server expires max 36000 min 600
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 1 pots
description ### External Dialling via BRI ###
preference 7
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/0
forward-digits all
dial-peer voice 2 pots
description ### External Dialling via BRI ###
preference 2
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/1
forward-digits all
dial-peer voice 9000 voip
description ** Outgoing calls to SIP **
preference 1
destination-pattern 9T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:99.234.56.78:5060
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
timers connect 100
sip-server ipv4:99.234.56.78
I used debugging commands to troubleshoot the calls.
2801(config-dial-peer)#
094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018
094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9
094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH
094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90
094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH
094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908
094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH
094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086
094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH
094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862
094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH
094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908621
094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH
094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086215
094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH
094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157
094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH
094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908621577
094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086215777
094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774T
094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL
094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397230
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397231
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:12 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397232
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:14 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397234
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I made some changes in the router configuration.
I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
The debugging is changed now. I can send and receive a respond from SIP server. But It shows an error: SIP/2.0 404 Not Found
Then it moves to ISDN line, and use this line to make a call.
102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774T
102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327416347
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 19412 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Call-ID: [email protected]
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
Content-Length: 0
103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=211
103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=20018
103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=0862157774
103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=NO_MATCH(-1)
103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
2801(config-dial-peer)#
Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
But it didn’t affect anything.
Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
Really stuck and don't know where to look at.
Any help will be highly appreciated.
Thanks.

Hi Dan.
Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
I use Cisco ASDM for ASA to make changes.
There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44) for a few ports.
Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
For NAT:
I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44
(AS TRANSLATED) UDP 5060
Because there is already translation for the Server.
Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.
116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:25 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505305
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:26 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505306
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:27 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505307
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505337
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I'll add Incoming dial-peer now.
Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
Appretiate your help.
Thanks a mill.

VoIP (SIP) configuration guide for Nokia 5800xm

hi.everybody i would like to ask how to configure this
VoIP (SIP) configuration for my n5800, because when i saw and follow this link http://www.elisanet.fi/craig/sipvoip/nokia_n97.html ,i only configure my SIP setting and not my net setting which i tried to find in my phone. can someone help me find this Net setting in my handset and can you put a link on how to call from phone to phone using this "VoIP" and is this free of charge. thank..more power

Hi, don't have an answer to your question, but Skype for S60.5 was announced today, works fine, will give Skype to Skype calls free, and low cost calls to non Skype numbers, but both Skype or Voip will incur data charges if you don;t have a data plan !
Good Luck
Skype.com/m with your phones browser will take you to Skype
If I have helped at all, a click on the White Star is always appreciated :
you can also help others by marking 'accept as solution'

How to Invoke Oracle Configurator via URL Outside the APPS Firewall

Hi Gurus,
We would like to invoke Oracle Configurator via URL same as Oracle iStore. Please let me know the process/steps to meet the requirement.
Thanks in Advance,
Venky.

There is no restriction that Oracle Configurator may only be executed from within an Oracle hosting application. Configurator may be invoked from any application that has the ability to call Configurator's UI servlet. As an example, Astec (now part of Emerson Network Power) has had a Configurator application they call their "Power Wizard" on their public website for nearly eight years (http://www.powerconversion.com/powerwizard/).
Venky, I would recommend you search My Oracle Support for articles containing 'configurator firewall' or 'configurator ssl', and then filter the results to just EBS articles. If what you find there is not helpful, opening a Support Request to get information more tailored to your particular scenario may be advisable.
Eogan

Sip configuration for Vyke

Hi
Can someone tell me where to find the Sip configuration for VYKE system?

Profile name: Arbitrary
Service profile: IETF
ACCESS POINT : [your wifi access point]
PUBLIC USER NAME: sip:[username]@sip.vyke.com
use compression: no
registration : “always on” or “when need”
SECURITY: NO
PROXY SERVER
PROXY SERVER: sip:sip.vyke.com
REALM: sip.vyke.com
USER NAME: :[sip_user]
password: [sip_pass]
allow loose routing: yes
transport type: UDP
PORT: 5060
REGISTRATION SERVER
REGISTRATION SERVER: sip:sip.vyke.com
realm: sip.vyke.com
user name: [sip_user]
password: [sip_pass]
transport type: UDP
PORT: 5060
N86 8mp: RM-485,0590552; Version 20.115 - 6/29/2009. - -- E71-2; RM-357; 0569371; Version 100.07.76 - 6/08/2008.
N8; N900

SIP configuration trouble

Hi,
On my gateways 5400, SIP configuration is :
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface Loopback6
bind media source-interface Loopback6
But on one of my 5400, line "bind control source-interface Loopback6" doesn't appear. When i implement it, i have no error message.
SIP calls doesn't complete (dead air). Dial-peer is well configured (calls completed on another gw).
Thanks for your help.
Phil

Compare the IOSes running on these AS5400s.
Are they different? :)
If they are, choose one that you consider as being most stable and make the other running it :)

Cisco UC5xx 8.6 Support for 99xx SIP phones using CCA 3.2.1

Hi Friends,
I was looking through posts here in the SMB commuity, the SWP 8.6 for UC5xx, the RN for CCA 3.2.1, and the OLH for CCA 3.2.1, and found a nice thread that will help anyone wanting to get a 9951 or 9971 SIP phone to operate on the UC5xx after upgrade to the Cisco IOS {15.1(4)M4} bringing CME 8.6 to Telephony Services and selection of the SIP 9-2-2 phone loads (included in the SWP) for 99xx.
My only inquiry for Cisco to check, would be why isnt this documented in the release notes,
https://supportforums.cisco.com/servlet/JiveServlet/previewBody/26979-102-1-67567/cca_3_2_2_relnotes.pdf
since CCA doesnt seem to add the 'load 99xx sip99xx.9-2-2' statement, the 'tftp-path flash:', followed by the 'create profile' under VOICE REGISTER GLOBAL?
If it is supposed to work, then I would alert you that it did not. After Upgrading CCA to 3.2.1 and then upgrading the UC5xx to SWP 8.6, the adminisrtrator manually adds the 99xx phone by entering MAC and Type under Configure> Telephony> Users/Extensions> Users and Phones: ADD button. Nothing special, just a normal extension on one button, Video Enabled, and a VM box created. This allows the phone to register just fine, but it doesnt automatically upgrade to 9-2-2 due to the missing bold commands above.
I think if this were a known defect, it would have been documented in the RNs, so I raise it to your attention.
Which operation should have added these commands?
Can you let us know if this is an anomaly or if everyone will encounter this?
Thanks kindly,
Steve

Yeah uh beleive me Steven, I have tried everything and every location to get these phones working and nothing did. Other people have the same issue (thread here somewhere) luckily mine are only out 1 hour others are out many hours.
Thanks,
Bob James

CMR SIP alias via Productvity tools

Recently added CMR and running into an issue with SIP Alias for video endpoints. When scheduled via the Webex portal, the SIP alias is correct ([email protected]). However when we schedule using Productivity Tools or via Jabber, it schedules the meeting fine but then assigns the video alias differently. Rather than users firstname.lastname, it assigns the meeting number as the video alias. So now its [email protected] I need to figure out how to create a consistent meeting invite.

Hi,
So your current situation is that you are using AnyConnect with Split-tunneling? So you can access the LAN through VPN and Internet through the users local Internet connection?
I wonder if the problem with Full-Tunnel + Internet traffic is due to problem with NATing the traffic from VPN Client pool to the "outside" interface IP?
I think that NAT could be done by the following configuration
object network VPN-POOL
subnet 192.168.0.200 255.255.255.248
nat (outside,outside) after-auto source dynamic VPN-POOL interface
EDIT: The "after-auto" should take the rule at the bottom of the NAT rules, but I'm still thinking should it interfere with the configurations even if it didnt have the parameter.
I've only done this in software 8.2 and below, where it was
global (outside) 1 x.x.x.x
nat (inside) 1 10.10.10.0 255.255.255.0
nat (outside) 1 192.168.0.200 255.255.255.248
But as I said I havent done this with the new software. I can probably test this tomorrow at work though.
- Jouni
EDIT2:
I'm not 100% sure but is the following NAT statement even needed at the current configuration?
nat (inside,outside) source static any any destination static NETWORK_OBJ_192.168.0.200_29 NETWORK_OBJ_192.168.0.200_29
EDIT3:
I also usually use a totally different network for the VPN-pool. Your current LAN seems to be 192.168.0.0/24 and the VPN pool is 192.168.0.200/29. Maybe the top NAT statements are causing problems with the VPN Client Internet traffic.
What I would try to do is change the VPN Client pool to some different network, do the NAT0 between LAN and POOL like you have in the current configuration (just with new VPN Pool) and use the PAT configuration mentioned earlier. Leave out the NAT statement in the "EDIT2" in this post.
Sorry this is alot of guessing from my part before I get to test it myself with a lab ASA.

UC560 SIP Trunk, IOS Jabber, MWI

Has anyone successfully been able to configure complete functionality of these on the UC560? Making IOS Jabber work, breaks sip trunk configuration and the MWI. I do currently have a case opened with TAC, but they have yet to find the solution. Currently the sip trunk is working as is the MWI, the IOS Jabber phone is registered, but can not make or receive phone calls.

Good morning
Hi John, thanks for using our forum, my name is Johnnatan and I am part of the Small business Support community. You can post your question in "Small Business Voice and Conferencing>SBCS - UC500" so you can have more feedback on your case, more users will see it there. You can move your post using the actions panel on the right.
Greetings,
Johnnatan Rodriguez Miranda.
Cisco Network Support Engineer.

SIP Configuration for W995

How can i configure and use SIP in sonny W995.
Basmaa

the standard software by W995 don't have any option to call via SIP!! I don't understand why this phone has such a Otption that you can not use

Importing configurations via Order Import in R12

Hi All,
We need some help with this issue we are facing while importing configurations ( PTO ) via order import process.
The scenario is like this : we have parent PTO Model Item : A and under this we have 2 Option Classes within the BOM: OC1 & OC2
Each of these option classes have 2 items each : option class - OC1 has ( Item B & Item C) while option class - OC2 has ( Item D & Item E)
Now in the OM interface tables we want to interface : PTO Model Item A and along with it the Item B and Item E ( one item from each of the option classes). Hence we populate the oe_lines_iface_all table the below mentioned fields
inventory_item, item_type_code , orig_sys_line_ref , top_model_line_ref, ship_from_org
A                     MODEL               1.0                        NULL                      S1
B                                                                            1.0                         S1
E                                                                            1.0                         S1
--Now when we run the Order Import we are getting errors : "org_sys_line_ref cannot be NULL" and errors - "item B and E are not found in the BOM of the model A"
--when we try to populate the orig_sys_line_ref columns values for Item B and E as let's say 2.0 and 3.0 , still we get the "items not found in the BOM of the model A error message". But when we check the BOM these component items are present there.
With the same Model Item A and Option class component items B and E we are able to create an order successfully within OM frontend screen and the BOM structure for the PTO Model A is also active and enabled.. This is a Published BOM for the PTO Model Item A
Could someone please help as to what additional columns need to be populated or what needs to be done in order to get this resolved.
Thanks

We referred the Oracle OM related white paper from metalink :
Oracle Order Management Suite White Papers (Doc ID 113492.1) - Importing Configurations

Package Configuration via SQL Server For a Reporting Queue

The scenario is a reporting queue for a web application. Currently when a user queues a report it stores the parameters in a table by queue ID. When that report is set to run the database creates a dtexec command with all the parameters. We use this
method for transferring the values the user selected to the SSIS packages.
I have used SQL Server package configurations before but they are usually a single table with a distinct ConfigurationFilter column. Is it possible to take the standard table layout for the SQL Server package configuration and add another column, say
QueueID. We send the QueueID to each package at run time and use this value to filter the table to the configuration specific to that report?

Here is a better break down of the current process:
1. A user goes to the web application and selects a report/parameters.
2. This will add data to [QUEUE] and [QUEUE_PARAMETERS] tables associated by the QUEUE_ID
3. The server has a windows service which looks for reports in the [QUEUE] table in pending status and gets the next one available and calls a stored procedure with that QUEUE_ID.
4. The stored procedure puts together a dtexec command through xp_cmdshell which contains all the user parameters from the [QUEUE_PARAMETERS] table.
The purposed process is to replace the self made [QUEUE_PARAMETERS] table with the package configuration table. This means we would only have to send the QUEUE_ID via the dtexec command and not all the other parameters. The problem is that if
I add QUEUE_ID to the package configuration table I am not sure how I would reference a specific QUEUE_ID from the SSIS package configuration dynamically. There does not appear to be an expression that can be applied to the package configuration in SSIS
which means the relationship must make the ConfigurationFilter field unique, even though it is not a primary key.

CME Extension Mobility, SIP configuration

Hi,
Need help with CME Extension Mobility with SIP Phones (7841). I'm using CME 10.5 and I configured the parameters below for extension mobility but the phones won't register right after I put the logout profile in the voice register pool.
They work normally when not in Extension Mobility though. Please help I need to deploy this to my customer soon.
hostname Router
boot-start-marker
boot system flash:c3900-universalk9-mz.SPA.154-3.M2.bin
boot-end-marker
no aaa new-model
no authentication logging verbose
ip dhcp excluded-address 192.168.1.1 192.168.1.20
ip dhcp excluded-address 192.168.1.254
ip dhcp pool Phones
network 192.168.1.0 255.255.255.0
default-router 192.168.1.254
option 150 ip 192.168.1.254
no ip domain lookup
ip cef
no ipv6 cef
multilink bundle-name authenticated
cts logging verbose
voice-card 0
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/1.10
bind media source-interface GigabitEthernet0/1.10
registrar server expires max 600 min 60
voice register global
mode cme
source-address 192.168.1.254 port 5060
max-dn 110
max-pool 110
load 7841 sip78xx.10-1-1SR1-4
time-format 24
date-format D/M/Y
service https
url authentication http://192.168.1.254/CCMCIP/authenticate.asp
tftp-path flash:
create profile sync 0002641841434163
voice register dn 1
number 6001
name Poh Huat - 6001
label Poh Huat - 6001
voice register dn 4
number 6005
name Coordinator - 6005
label Coordinator - 6005
voice register pool 1
logout-profile 100
busy-trigger-per-button 2
id mac 547C.69D6.1AB6
type 7841
voice register pool 4
logout-profile 100
busy-trigger-per-button 2
id mac 547C.69D6.1A2F
type 7841
voice logout-profile 100
pin 1234
user 6000 password 12345
number 6000 type normal
speed-dial 1 999 label "EMERGENCY"
voice user-profile 1
pin 12345
user richard password richard
number 6001 type normal
speed-dial 1 996506901 label "Richard"
voice user-profile 2
pin 12345
user 6005 password 12345
number 6005 type normal
license udi pid C3900-SPE100/K9 sn FOC16145MQA
license boot module c3900 technology-package uck9
username xtra privilege 15 secret 5 $1$STRs$Qsuesm8dF23Okof.vRyf5.
redundancy
ip ftp username xtra
ip ftp password xtra2006admin
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address dhcp
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
interface GigabitEthernet0/1.10
encapsulation dot1Q 10 native
ip address 192.168.1.254 255.255.255.0
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
no ip http secure-server
ip http path flash:
nls resp-timeout 1
cpd cr-id 1
tftp-server flash:PHONES/sip78xx.10-1-1SR1-4.loads alias sip78xx.10-1-1SR1-4.loads
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
gatekeeper
shutdown
telephony-service
authentication credential 6000 12345
em keep-history
max-ephones 110
max-dn 110
service phone webAccess 0
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 05 2015 15:50:52
I have turned on debug ip http all and debug voice em-profile on and right after I entered the logout profile 100 under pool i get the following logs.
Router(config-register-pool)#
Mar 5 16:15:28.299: Thu, 05 Mar 2015 16:15:28 GMT 192.168.1.21 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query = locale=English_United_States&name=SEP547C69D61A2F
Mar 5 16:15:28.299:
Mar 5 16:15:28.299: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:28.299: SIP phone 4 found with contact IP address 192.168.1.21
Mar 5 16:15:33.363: Thu, 05 Mar 2015 16:15:33 GMT 192.168.1.21 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query = locale=English_United_States&name=SEP547C69D61A2F
Mar 5 16:15:33.363:
Mar 5 16:15:33.363: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:33.363: SIP phone 4 found with contact IP address 192.168.1.21
Mar 5 16:15:37.539: Thu, 05 Mar 2015 16:15:37 GMT 192.168.1.21 /CMEserverForPhone/extensionmobility ok
Protocol = HTTP/1.1 Method = GET
Mar 5 16:15:37.539:
Mar 5 16:15:37.539: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:37.539: SIP phone 4 found with contact IP address 192.168.1.21
After this the phones are still not registering, I'm suspecting it is the url authentication command, as i can't put the application-name and password after the command, any suggestions would be appreciated. THanks in advance.
-richard

Try adding:
voice register global
url authentication http://192.168.1.254/CCMCIP/authenticate.asp secretname psswrd
if still doesn't work try to compare your config with the reference guide here:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmemobl.html#pgfId-1163414
-Terry
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How to retrieve Nexus 7010 et 5596 configuration via Snmp ?

Hello all,
I want to know how to retrieve the complete configuration for a Nexus via the snmpwalk or snmpget commands...
What mib's is needed ?
Thanks for the feedback

Hi more information, i execute the scrip below :
#!/bin/sh
# Script to demonstrate the SNMP commands necessary to backup the Cisco Nexus Switches
COMMUNITY="RW community"
HOST="Ip @"
TFTPSERVER="srv @"
FILENAME="nex7000.cfg"
RND="10"
snmpset -v 2c -c$COMMUNITY $HOST ccCopyProtocol.$RND i 1 ccCopySourceFileType.$RND i 4 ccCopyDestFileType.$RND i 1 ccCopyServerAddress.$RND a $TFTPSERVER ccCopyFileName.$RND s $FILENAME ccCopyEntryRowStatus.$RND i 4
sleep 5
snmpget -v 2c -c$COMMUNITY $HOST ccCopyState.$RND
sleep 5
snmpget -v 2c -c$COMMUNITY $HOST ccCopyState.$RND
sleep 5
snmpget -v 2c -c$COMMUNITY $HOST ccCopyState.$RND
sleep 5
snmpget -v 2c -c$COMMUNITY $HOST ccCopyState.$RND
snmpset -v 2c -c$COMMUNITY $HOST ccCopyEntryRowStatus.$RND i 6
When i launch it i have this message on the server :
./testnexus.sh
ccCopyProtocol.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyProtocol)
ccCopySourceFileType.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopySourceFileType)
ccCopyDestFileType.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyDestFileType)
ccCopyServerAddress.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyServerAddress)
ccCopyFileName.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyFileName)
ccCopyEntryRowStatus.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyEntryRowStatus)
ccCopyState.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyState)
ccCopyState.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyState)
ccCopyState.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyState)
ccCopyState.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyState)
ccCopyEntryRowStatus.10: Unknown Object Identifier (Sub-id not found: (top) -> ccCopyEntryRowStatus)
When i launch a debug snmp all, i have no error, i just see that the packet that comes it's accepted by the ACL
Thanks for your feedback

UC560 SIP Configuration via CCA

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