2650XM dial-peer using Trunks

Hi,
I got a customer with a 2650XM IOS vers. 12.4(19b) (c2600-adventerprisek9_ivs-mz.124-19b), I cannot use ports only Trunks and I cannot get the dial-peers to work. I programmed the Controller with the PRI and D-channel with a Trunk label but not voice-port is created.
I had to replace his router with a temporary 2821 and everything is Ok but of course I need to recover the 2821, his router was working before with another solution, not Cisco, so they didn't want to buy a new one and a PRI card.
I am guessing I don't understand the whole TRUNK programming for Dial-Peers, any help will be appreciated, below are the pieces of config and debug relevant to this.
Regards,
Hiram
CONFIG
trunk group PSTN
description PRI Line to PSTN
controller T1 1/0
framing esf
linecode b8zs
pri-group timeslots 1-24
description PRI T1 for PSTN
interface Serial1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-dms100
isdn incoming-voice voice
isdn supp-service name calling
trunk-group PSTN
no cdp enable
dial-peer voice 1 pots
incoming called-number .
trunk-group-label source PSTN
direct-inward-dial
dial-peer voice 201 pots
description Primary Outbound Dial Peer for 10 Digits - Local Services
preference 1
destination-pattern 8305[2-9]......
trunk-group-label target PSTN
prefix 305
dial-peer voice 1000 voip
preference 1
destination-pattern 1...
voice-class codec 5
voice-class h323 5
session target ipv4:192.168.204.21
incoming called-number 8T
dtmf-relay h245-alphanumeric
dial-peer voice 1011 voip
preference 1
destination-pattern 1...
voice-class codec 5
voice-class h323 5
session target ipv4:192.168.204.21
dtmf-relay h245-alphanumeric
DEBUG
HQ1VGW1#
*Mar 2 07:57:46.828: //-1/80D9DD911300/DPM/dpAssociateIncomingPeerCore:
Calling Number=1000, Called Number=83059998002, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 2 07:57:46.828: //-1/80D9DD911300/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
*Mar 2 07:57:46.832: //-1/80D9DD911300/DPM/dpAssociateIncomingPeerCore:
Calling Number=1000, Called Number=83059998002, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 2 07:57:46.832: //-1/80D9DD911300/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
*Mar 2 07:57:46.848: //-1/80D9DD911300/DPM/dpMatchPeersCore:
Calling Number=, Called Number=83059998002, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 2 07:57:46.848: //-1/80D9DD911300/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=83059998002
*Mar 2 07:57:46.848: //-1/80D9DD911300/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Mar 2 07:57:46.848: //-1/80D9DD911300/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
HQ1VGW1#

Thanks for responding.
HQ1VGW1#sh diag
Slot 0:
C2650XM 1FE Mainboard Port adapter, 2 ports
Port adapter is analyzed
Port adapter insertion time 17:09:56 ago
EEPROM contents at hardware discovery:
Hardware Revision : 3.0
PCB Serial Number : FFFF
Part Number : 73-7755-04
RMA History : 00
RMA Number : 0-0-0-0
Board Revision : A0
Deviation Number : 0-0
Product (FRU) Number : C2650XM-1FE
EEPROM format version 4
EEPROM contents (hex):
0x00: 04 FF 40 03 6E 41 03 00 C1 0B FF FF FF 46 46 46
0x10: 46 FF FF FF FF 82 49 1E 4B 04 04 00 81 00 00 00
0x20: 00 42 41 30 80 00 00 00 00 FF FF FF FF FF FF FF
0x30: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x40: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x50: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x60: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x70: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
WIC Slot 0:
FT1 BT8360
Hardware revision 1.3 Board revision E0
Serial number 25639147 Part number 800-03279-04
FRU Part Number WIC-1DSU-T1=
Test history 0x0 RMA number 00-00-00
Connector type Wan Module
EEPROM format version 2
EEPROM contents (hex):
0x20: 02 11 01 03 01 87 38 EB 50 0C CF 04 00 00 00 00
0x30: 70 00 00 00 02 08 27 01 FF FF FF FF FF FF FF FF
Slot 1:
CT1 (CSU) Port adapter, 1 port
Port adapter is analyzed
Port adapter insertion time 17:09:55 ago
EEPROM contents at hardware discovery:
Hardware revision 1.1 Board revision B0
Serial number 29805542 Part number 800-01228-05
FRU Part Number NM-1CT1-CSU=
Test history 0x0 RMA number 00-00-00
EEPROM format version 1
EEPROM contents (hex):
0x00: 01 26 01 01 01 C6 CB E6 50 04 CC 05 00 00 00 00
0x10: 58 00 00 00 04 05 08 00 FF FF FF FF FF FF FF FF
0x20: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x30: FF FF FF FF FF FF FF FF FF FF FF FF
HQ1VGW1#
HQ1VGW1#sh isdn status
Global ISDN Switchtype = primary-dms100
ISDN Serial1/0:23 interface
dsl 0, interface ISDN Switchtype = primary-dms100
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x807FFFFF
Number of L2 Discards = 0, L2 Session ID = 2
Total Allocated ISDN CCBs = 0
HQ1VGW1#
Regards,
Hiram

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Best practice about dial-peer creating when using analog lines

Hi,
I am trying to find out what is the best practice when creating dial-peer for analog lines on CME, should I use trunk group or create separate dial-peer for each FXO ports? If I use trunk group, is there any advantage ( lesser dial-peer) or disadvantage?
Thanks!

The advantage of trunk groups is that a single dial peer can point to for instance PSTN, rather then multiple dialpeers, with varying preference, each pointing to a separate FXO. Funtionally I can't see much difference. So I guess it also comes down to personal preference.
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How monitor the use of dial-peer (Connect Time)

How monitor multiple dial-peer (Connect Time) ??
When this reaches the defined time accumulated connected (example: Connect Time = 10000) shutdown the dial-peer.
UC520#show dial-peer voice 2 | beg Connect
Connect Time = 1430, Charged Units = 0,
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This should be doable with an EEM policy. If I understand you correctly you want to watch for connect time to reach 10000 when it does you want to shutdown the dial peer.
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The first part you would do with a watchdog EEM policy which periodically runs and checks the output of the show dial-peer voice 2 command.
The second part you probably would use a second EEM script to trigger on a cron timer set for the 15th day of the month, and have it run the commands you mentioned.
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Using answer-address in dial-peer when it has incoming called number

Hello,
We would like to use an inbound dial peer that will match by the Calling Number using answer-address.
In our current configuration we already have inbound dial peers with "incoming called number" for Fax2Mail services.
We want to use the calling number for matching to a different dial peer, without the "fax detect" service.
As we notice, as soon as he matching the incoming called number, even if have identical dial peer with the same incoming called number, he stop the matching processes and the gateway ignore the answer-address.
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Thanks

Hello,
in the Inbound Dial-peer matching process, the 'incoming called-number' has the highest priority over asnwer- address & destination pattern. So if you want to match a particular inbound dial-peer based on calling number, add only asnwer-address and don't configure incoming-called number on the same dial-peer.
FYI: Understanding the dial-peer matching process
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html?referring_site=smartnavRD#topic3
//Suresh
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Incoming Dial-peer

I have a SIP trunk set up and can make outgoing calls fine. However, at the moment, i'm having a little trouble getting the incoming calls to be recieved on my ephone-dn.
When an incoming call is recieved I get the following from "debug ccsip all"
The Call Setup Information is:
Call Control Block (CCB) : 0x4838BFD4
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +4418338xxxxx
Called Number : 4418336xxxxx
Source IP Address (Sig ): 83.x.x.x
Destn SIP Req Addr:Port : 87.x.x.x:5060
Destn SIP Resp Addr:Port : 87.x.x.x:5060
Destination Name : 87.x.x.x
Aug 25 08:24:38.529: //2100/FB9CA81483B9/SIP/Call/sipSPIMediaCallInfo:
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Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
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Dtmf-relay Payload : 0
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Source IP Port (Media): 18736
Destn IP Address (Media): 87.x.x.x
Destn IP Port (Media): 12272
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I believe my problem lies with me incorrectly setting up my dial-peer. This is what i currently have.
dial-peer voice 9999 voip
description Incoming call via SIP
translate-outgoing called 3
destination-pattern 4418336xxxxx
translation-rule 3
Rule 0 ^4418336xxxxx 2000
2000 is my extension on CME4.0 and i've x'd up IPs and phone numbers for privacy.
I would have thought that translation-rule would replace the called number with my extension, which would in turn cause my extension to ring. No such luck though.
I can get the call to be put through to a normal phone via a pots dial-peer with the following dial-peer
dial-peer voice 9999 pots
destination-pattern 4418336xxxxx
port 0/1/1
Thanks

Still pulling my hair out with this. I have tried your suggestion with the following setup:
voice translation-rule 10
rule 1 /^441833612345/ /2000/
voice translation-profile incSIP
translate called 10
dial-peer voice 1 voip
translation-profile incoming incSIP
incoming called-number .
Then with "debug voice dialpeer" activated, I get the following.
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=441833612345, Called Number=441833600026, Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=441833612345
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
Aug 25 14:29:25.219: //-1/F1237AF8804E/DPM/dpAssociateIncomingPeerCore:
Calling Number=+441833912345, Called Number=441833600026, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:25.219: //-1/F1237AF8804E/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
Aug 25 14:29:27.663: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=441833612345, Called Number=441833600026, Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:27.663: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=441833612345
Aug 25 14:29:27.667: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Aug 25 14:29:27.667: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
Aug 25 14:29:27.667: //-1/F29902FC8050/DPM/dpAssociateIncomingPeerCore:
Calling Number=+441833912345, Called Number=441833600026, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:27.667: //-1/F29902FC8050/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1

Dial-Peer matches but fails to call out

Hello,
Am trying to get my CME configured for Callcentric. I have both an inbound and an outbound plan.
With my dial-peers configured for standard 11-digit and 10-digit dialing, calls go to fast busy after all digits except the last two are dialed. Debug shows a dial-peer match initially, then states no match and the call fails. If I change the destination pattern to match my cell phone number exactly, I can dial all the digits but the call still fails. Anyone have a suggestion?
Here are my dial peers:
dial-peer voice 700 voip
description SIP Trunk - Incoming
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:callcentric.com
incoming called-number .%
dial-peer voice 701 voip
description SIP Trunk - Outgoing 3-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 9[2-8]11
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
dial-peer voice 702 voip
description SIP Trunk - Outgoing 11-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 91[2-9].......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
dial-peer voice 703 voip
description SIP Trunk - Outgoing 10-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 9[2-9].......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
no vad
And here is the debug associated with a call:
*Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=7018, Called Number=, Voice-Interface=0x4A4AE7B0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20009
GMIT-VOICEROUTER01#
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=91, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=91
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Numbe
GMIT-VOICEROUTr=, Called Number=912, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=912
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=91207, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=91207
*Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=912072, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=912072
*Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.542: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120722, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=91207227, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=91207227
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=912072277, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=912072277
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:32.606: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=702
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=702
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=9120722776, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=9120722776, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=702
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=702
*Dec 26 22:26:33.574: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=91[2-9]......., Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.578: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt
*Dec 26 22:26:36.374: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=7018$, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:36.378: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules AttemptER01#

Translation profile:
voice translation-rule 3
rule 1 /^7../ /2072267262/
voice translation-rule 4
rule 1 /^9$1....$/ /\1/
rule 2 /^9207$...$/ /\1/
rule 3 /^9$011.*$/ /\1/
rule 4 /^9$[2-9]11$/ /\1/
voice translation-profile SIP_1
translate calling 3
translate called 4
Here is debug ccsip messages:
*Dec 27 14:10:16.598: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
Max-Forwards: 69
Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
CSeq: 938331054 OPTIONS
Organization: MetaSwitch
Supported: resource-priority, 100rel
Content-Length: 0
Contact:
To:
*Dec 27 14:10:16.606: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
To:
GMIT-VOICEROUT166>;tag=F1B5120-18BD
Date: Fri, 27 Dec 2013 14:10:16 GMT
Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 938331054 OPTIONS
Supported: 100rel,resource-priority,replaces,sdp-anat
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 172
v=0
o=CiscoSystemsSIP-GW-UserAgent 4484 7548 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 66.55.220.166
ER01#
GMIT-VOICEROUTER01#
*Dec 27 14:10:34.834: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2066961728-1849102819-2185007278-567139419
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, B
GMIT-VOICEROUTER01#YE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1388153434
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297
v=0
o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 19258 RTP/AVP 18 101 19
c=IN IP4 66.55.220.166
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
*Dec 27 14:10:34.906: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
f: "Server Room" [email protected]>;tag=F1B9854-8A5
t: [email protected]>
i: [email protected]
CSeq: 1
GMIT-VOICEROUT01 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="3755ae79fd668c2035ebb90cdc12d030", opaque="", stale=TRUE, algorithm=MD5
l: 0
*Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2066961728-1849102819-2185007278-567139419
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: 1388153434
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772882353",realm="callcentric.com",uri="sip:[email protected]:5080",response="cbac03a76a23b6a35ebbee966c00a577",nonce="3755ae79fd668c2035ebb90cdc12d030",opaque="",algorithm=MD5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297
v=0
o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 19258 RTP/AVP 18 101 19
c=IN IP4 66.55.220.166
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
*Dec 27 14:10:34.990: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
f: "Server Room" [email protected]>;tag=F1B9854-8A5
t: [email protected]>
i: [email protected]
CSeq: 102 INVITE
l: 0
*Dec 27 14:10:35.002: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Here is debug voip ccapi inout:
GMIT-VOICEROUTER01#debug voip ccapi inout
voip ccapi inout debugging is on
GMIT-VOICEROUTER01#
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=7018
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-lastrdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x4A4AE7B0, Call Info(
   Calling Number=7018,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE,
   Incoming Dial-peer=20009, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
GMIT-VOICEROUT, Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
   In: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
   Out: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: :cc_get_feature_vsa malloc success
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: cc_get_feature_vsa count is 1
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234808,feature_id:151
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown))
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
   Event=0x49A103B8
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/ccCallSetContext:
   Context=0x4C5A319C
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 12898 with tag 20009 to app "_ManagedAppProcess_Default"
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccCallSetupAck:
   Call Id=12898
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_api_set_transfer_info:
   Transfer Number=, Transfer Reason=0x0
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=TRUE, Tone=Dial Tone,
   Tone Direction=Network, Params=0x0, Call Id=12898
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=-1000(ms), Inter Digit Timeout=-1000(ms)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=10000(ms))
*Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x3262, digit_event=0x1, enable=TRUE, consume=FALSE)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=12898
*Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
*Dec 27 14:10:56.650: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=9, DigitBeginFlags=0x0,
   Rtp Timestamp=0x9D41D0, Rtp Expiration=0x0
*Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=9, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:56.970: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=1, DigitBeginFlags=0x0,
   Rtp Timestamp=0x9DBED0, Rtp Expiration=0x0
*Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=1, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.290: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
   Rtp Timestamp=0x9E3BD0, Rtp Expiration=0x0
*Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=2, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.610: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=0, DigitBeginFlags=0x0,
   Rtp Timestamp=0x9EB8D0, Rtp Expiration=0x0
*Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=0, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.890: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
   Rtp Timestamp=0x9F35D0, Rtp Expiration=0x0
*Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=7, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
   Rtp Timestamp=0x9FB2D0, Rtp Expiration=0x0
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=2, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
   Rtp Timestamp=0xA02FD0, Rtp Expiration=0x0
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=2, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
   Rtp Timestamp=0xA0ACD0, Rtp Expiration=0x0
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=7, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
   Rtp Timestamp=0xA129D0, Rtp Expiration=0x0
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=7, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=6, DigitBeginFlags=0x0,
   Rtp Timestamp=0xA1A6D0, Rtp Expiration=0x0
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
   Source Call Id=12898, Digit=6, Duration=100,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
*Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x3262, digit_event=0x0, enable=FALSE, consume=FALSE)
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=12898
*Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=702, Params=0x4C5A0BDC, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
   In: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
   Out: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
   Destination Pattern=91[2-9]......., Called Number=120722776, Digit Strip=FALSE
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
   Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=120722776(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Server Room
   Account Number=, Final Destination Flag=FALSE,
   Guid=8912F77B-6E37-11E3-8243-90AE21CDDC5B, Outgoing Dial-peer=702
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=20722672628
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=120722776
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-lastrdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
*Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x48C27BD0, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=20722672628,(Calling Name=Server Room)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=120722776(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=702, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: :cc_get_feature_vsa malloc success
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: cc_get_feature_vsa count is 2
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234584,feature_id:152
*Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccCallSetContext:
   Context=0x4C5A0B8C
*Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=702
*Dec 27 14:10:59.038: //12899/8912F77B8243/CCAPI/cc_api_call_proceeding:
   Interface=0x48C27BD0, Progress Indication=NULL(0)
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
   Cause Value=57, Interface=0x48C27BD0, Call Id=12899
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
*Dec 27 14:10:59.270: //12898/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=12899
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
   Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=57)
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
   Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x48C27BD0, Tag=0x0, Call Id=12899,
   Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:10:59.274: :cc_free_feature_vsa freeing 4C6D58D0
*Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:10:59.274: vsacount in free is 1
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
   Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
   Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
*Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x4A4AE7B0, Tag=0x0, Call Id=12898,
   Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:11:02.250: :cc_free_feature_vsa freeing 4C6D59B0
*Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:11:02.250: vsacount in free is 0ER01#

Site to Site calling issue - Cisco 2911 Dial Peer Configuration

My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party.
At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue. We cut that site over to our service and the issue started right up. I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX. Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
Cisco equipment/Dialpeer config below ........
co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
dial-peer voice 100 voip
description --- VoIP Dial-Peer ---
translation-profile outgoing 7digit
huntstop
preference 1
service session
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 99
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
ip qos dscp af41 signaling
no vad
dial-peer voice 150 voip
permission none
description 900 block
huntstop
destination-pattern 1900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 151 voip
permission none
description 900 block
huntstop
destination-pattern 900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 101 pots
description --- INCOMING Calls from PBX ---
incoming called-number .T
direct-inward-dial
dial-peer voice 1001 pots
description --- Calls to the PBX ---
preference 3
destination-pattern .T
port 0/0/1:23
forward-digits 4
Here is some ISDN debug information
BAD CALL
Protocol Profile = Networking Extensions
0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
Component = Invoke component
Invoke Id = 66
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, ''6551''
Plan:Unknown, Type:Unknown
Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
GOOD CALL
Protocol Profile = Networking Extensions
0xA116020144020100800E475245454E204D4F554E5441494E
Component = Invoke component
Invoke Id = 68
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, 'XXXX''
Plan:Unknown, Type:Unknown
Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
Channel ID i = 0xA98381
Exclusive, Channel 1

I done the configration via CCA and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
dial-peer voice 2100 voip
corlist incoming call-internal
description **CCA*INTERSITE inbound call to SITE 1
translation-profile incoming multisiteInbound
incoming called-number 82...
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
dial-peer voice 2101 voip
corlist incoming call-internal
description **CCA*INTERSITE outbound calls to SITE2
translation-profile outgoing multisiteOutbound
destination-pattern 81...
session target ipv4:192.168.50.1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
no dial-peer outbound status-check pots

Outbound Dial-Peer from CME to UC540 not working

Dear Experts,
We have a HQ UC560 and new branch with 2811 router. These sites connected via VPN using fortigate.The connectivity between the sites is up and we are able to ping both the sites and the voice networks successfully.
I have configured the dial-peers on both the sites. The calls from HQ to the local branch are successful without any problem but when we dial from branch to the HQ, we get a fast busy signal. Below is the dial peer config
HQ -
dial-peer voice 300 voip
destination-pattern 3..
session target ipv4:192.168.110.1
dtmf-relay h245-alphanumeric
no vad
Branch -
dial-peer voice 800 voip
destination-pattern 8..
session target ipv4:192.168.201.2
dtmf-relay h245-alphanumeric
no vad
Csim results from Branch -
csim start 891
csim: called number = 891, loop count = 1 ping count = 0
csim err csimDisconnected recvd DISC cid(786)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0
Kindly please advise. thanks.

Hi, It is as suspected Toll Fraud App who rejected the call from BR site.
1076043: Oct 11 14:36:29.759: //282614/B639957688BC/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=308(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=807(TON=Unknown, NPI=Unknown))
1076047: Oct 11 14:36:29.763: //282614/B639957688BC/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 282614 with tag 300 to app "_ManagedAppProcess_TOLLFRAUD_APP"
1076048: Oct 11 14:36:29.763: //282614/B639957688BC/CCAPI/ccCallDisconnect:
   Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
you need to add BR GW IP address (192.168.110.1) to under 'voice service voip> ip address trusted list' as given below.
voice service voip
ip address trusted list
ipv4 192.168.110.1
For you reference: https://supportforums.cisco.com/document/46566/understanding-toll-fraud-enhancements-1512t

IP address based dial-peer matching

Hi,
is it possible to define a rule for incoming voip dial-peer matching based on the source IP address of originating VOIP gateway ?
I need it for differentiate outgoing POTS lines based on the source gateway. Any ideas how to do it ?
Thanks,
Vladimir

HI
Yes is it, to using voice source-group features..
Simple examble:
voice source-group vladimir
access-list 50
disconnect-cause call-reject
translation-profile incoming 50
voice translation-profile 50
translate called 50
voice translation-rule 50
rule 1 /^1$.*$/ /6501\1/
access-list 50 permit 10.0.0.0 0.0.0.255
dial-peer voice 52 pots
destination-pattern 650T
progress_ind alert enable 8
progress_ind progress enable 8
direct-inward-dial
port 7/0:D
rgds,
Ismo

Selecting dial peer based on the ip address of the incoming leg

hello all
i have configured 2 dial peers. i want to select dial peer 1 one the call comes from 1.2.3.4 and select dial peer 2 when the call comes from 3.4.5.6 i am not sure how to search for the same on the internet. please can anyone help me out with this..

HI
You can use voice source group feature with ACL and some prefixes, Below is an example of such an configuration..
access-list 1 permit 1.2.3.4 0.0.0.255
access-list 2 permit 3.4.5.6 0.0.0.255
voice source-group 1234
access-list 1
disconnect-cause invalid-number
translation-profile incoming 1
voice source-group 3456
access-list 2
disconnect-cause invalid-number
translation-profile incoming 2
voice translation-profile 1
translate called 1
voice translation-profile 2
translate called 2
voice translation-rule 1
rule 1 /^1$.*$/ /81\1/ type any subscriber plan any isdn
voice translation-rule 2
rule 1 /^1$.*$/ /71\1/ type any subscriber plan any isdn
dial-peer voice 1 voip
destination-pattern 8T
dial-peer voice 2 voip
destination-pattern 7T
rgds,
Ismo

Cisco dial-peer path selection with "preference"

Hi everybody,
for a test lab environment i'm testing the integration between cisco voice gateway 3925 and third party voice gateway by means of isdn PRI.
here the connection schema:
PSTN (emulated)-----> port0/0/0-Cisco3925-port0/0/1 <------- Third party Voice Gateway
                                                              |     (ethernet)
                                                      Cisco CUCM (172.23.112.20)
in brief:
- i'm emulating PSTN with a cisco voice gateway, this gateway is connected to cisco3925's port 0/0/0.
- cisco3925's port 0/0/1 is connected to Third party Voice Gateway.
- cisco 3925 speaks with Cisco CUCM in H323.
Now let's go for an incoming call from the PSTN when 3925 has no connection to CUCM, with called number 321672711 (321672... is the GNR of the site):
1. inbound: dial-peer 110 finds match so the called number is transformed to 591711 (it is a DN not registered to SRST cisco gateway)
2. outbound: i expect dial-peer 100 to be matched, because 172.23.112.20 is no more reacheable. From the show call active voice dial-peer 1 is matched as the attached. I need to set preference 1 in dial-peer 100 because when WAN is UP i don't want dial-peer 100 to be matched (and it works). But when WAN is down dial-peer 100 must match. If i remove preference 1, dial-peer 100 finds match; but for correct path selection i cannot remove it.
What am I forgetting?
thanks for support
voice translation-rule 1
rule 1 /^321672/ /591/
voice translation-profile ENTRANTE
translate called 1
(translate calling omitted)
dial-peer voice 1 voip
description Inbound per USCENTI - Outbound per ENTRANTI
corlist incoming CSSSRSTInternazionali
tone ringback alert-no-PI
destination-pattern 591...
session target ipv4:172.23.112.20
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 100 pots
preference 1
translation-profile outgoing NOMIG
destination-pattern 591...
port 0/0/1:15
dial-peer voice 110 pots
corlist incoming CSSSRSTInternazionali
description Inbound per ENTRANTI
translation-profile incoming ENTRANTE
incoming called-number 321672...
direct-inward-dial
port 0/0/0:15

Hello Marco,
There could be two possibilities:
1. To avoid dial-peer 1 being selected in the dialplan match, when gateway is trying to route the call, you can configure ICMP Probe , which would mark dial-peer as down, in case of WAN failure. So call will use dial-peer 100, automatically, as that will only be an possible match.
Here is document , in case you are interested in ICMP Probe:
http://www.cisco.com/c/en/us/td/docs/ios/voice/command/reference/vr_book/vr_m3.html#wp1397581
2. Ideally default dial-peer hunting mechanism is, Longest - Preference - Random , so as both the dial-peer has same destination pattern, in terms of specific digits and number of wild cards. So it should be looking as preference value of two possible matches, so in this test dial-peer 1 would win. Router will try to route the call using that dial-peer, if fails it should automatically fall back to dial-peer 100 as next choice.
But please note that it will still use dial-peer 1 at first attempt, as dial-peer status is not linked to interface status or WAN status. To verify this theory , you can remove session target command, and you will see that dial-peer 1, is not even selected in match, that's because removing session target command, will mark is as DOWN for outgoing status.
Taking below said debugs would help further, in case configuring ICMP probe is not viable option.
debug voip ccapi inout ( it will help understand , dial-peer match and hunting process ).
debug voip dialpeer inout
Hope that helps.

Multiple incoming dial-peer matched

Dear,
Could someone, tell me why after matching an incoming-peer on "answer-address" system matches for same call another incoming dial-peer based a "destination-pattern" ... shouldn't the system start looking for outbound dial-peer once first incoming diapeer matched?
046565: Feb 3 17:56:57: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=360
046566: Feb 3 17:56:57: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=1660, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
046567: Feb 3 17:56:57: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=360
046568: Feb 3 17:56:57: //-1/4C84D3B79834/DPM/dpAssociateIncomingPeerCore:
   Calling Number=1660, Called Number=+32495243137, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
046569: Feb 3 17:56:57: //-1/4C84D3B79834/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
Dial-peers:
dial-peer voice 360 voip
corlist incoming toBRICall
description ** AGENT COR PEER **
translation-profile outgoing E164IncomingSIP
preference 2
voice-class codec 1
session protocol sipv2
session target ipv4:10.161.10.170
dtmf-relay rtp-nte
fax rate disable
ip qos dscp cs5 signaling
no vad
destination-pattern 166.
answer-address 166.
dial-peer voice 1 voip
session protocol sipv2
incoming called-number .
dtmf-relay sip-notify
codec g711alaw
no vad
Thank you.
Best regards.
Marc

Hi
Dial peer matching (inbound and outbound) has a list of attributes that are used in a specific order.
For inbound peers, 'incoming called-number' has top priority, so the other command you set are not checked as this 'incoming called-number .' matches any called number!
Destination-pattern is the lowest priority match for an inbound peer, and answer-address is lower priroty than incoming called-number.
See this doc for the process in detail:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic1
Aaron HarrisonPrincipal Engineer at Logicalis UK
Please rate helpful posts...

As5350 one voip dial-peer to 2 IP address?

Is there any way to configure AS5350 so that one voip dial-peer can point to 2 different Ip address, one for primary, one for backup? so when primary is down, the traffic can be sent to backup automatically? BTW, I do not have GK. those 2 IP for redundant softswitch.

Hi,
your solution is very simple. Use two dial-peers pointing to the same destination-pattern (prefix/phone number). Set the appropriate IP address to each of the dial-peers. Then configure their preference. The one with the higher preference will be used as a primary and the other - as a backup.

Can't create pots dial-peer on CME router

I just acquired a cisco 2650, which i upgraded to "12.3 enterprise plus" IOS. i have CME configured on this router, with 2 7960s registered and working just fine. i add an NM-2V module with a 2FXS and a 2FXO cards into each of the NM slots. when i ty to configure the pots dial peers, the command parser doesn't show 'pots' as a valid dial-peer type:
confi t
dial-peer voice 1 pots
i get an error, and when i parse, only 'voip' is enabled as a valid dial-p type.
what i am doing wrong? is there any commands i am missing, in order to activate the pots? yet, when i do a "show voice port summary" i can clearly see the fxo ports in dormant stante, and when i plug an analog phone to it, i can hear the ringtone upon picking up.
please help.
regards
T.

i figured it out, finally. for some reason,if you issue "dial-peer voice 1", the only option available after integer 1 is "Voip". but when i pick any other integer >=2, then i get the option for either "voip" or "pots". so the bottom line is that i should have NOT used 1 for the dial-peer number.
why that is? go figure. i am aware of the defauld null dial-peer, but this one got me scratching my head. i can't say for sure that this worked for me in the past, but if this has been in the IOS for some time and i am only running into it now, then i guess my learning process have lots to look forward to :)
thanks for looking into this with me.
Regards,
T.

Fast response in dial-peer

I have a dial-peer match the string 10 and another matching string 100.
everytime I dial 100 it matches the dial-peer for 10 and rings before i even complete the dial.
is there anyway i can let the router wait to collect all digits before matching with a dial-peer?
PLS HELP!!!

Routers do not have the ability to see if a longer destination pattern will also match. As soon as the router finds a dialpeer that matches, the call is routed to the session target or port.
There are a couple ways to deal with this.
1) Have the shorter number a different first digit, e.i. 20, so that it does not match before the 100 pattern.
2) Make both patterns three digits long and make each unique.
3) Depending on your needs, you might be able to use the "T" to make the 10 pattern to wait for the length of the inter-digit timeout. This would mean that anyone dialing 10 will wait 10 seconds or so before the call routes.
A useful command for checking which dial peer the router will use is "Show Dialplan number XXX" XXX is the number that is being dialed.
Hope this helps.
Toby

2650XM dial-peer using Trunks

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