NEED GUIDANCE ON UNDERSTANDING DIAL-PEERS AND TRANSLATIONS-PROFILES....

[adsuther1]

Folks,

I need guidance on understanding the Dial-peers and Translations-profiles.

I must have read the help-guide 10 or more times;but the sequence of how the router process the dial-peers and translation-rules

is just not working for me.

Can anyone help me with this?  What can i do to get better understanding; I have a CME system i'm using for practice: but all the rules and sequences are not coming together for me.

HELP,

Anthony

[heathrw]

Have you read these:

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/64020-number-voice-translation-profiles.html

Also if you explain your configuration and what you are trying to do. ie send calls from an IP Phone to PSTN over ISDN, etc maybe able to provide some more assistance.

[adsuther1]

Heathrw,

thanks for the reply. Here is my first issue:

1. i just setup a cme configuration with one router; my cme router; i have two IP phones connected to a switch and cme router connected to the switch. IP phone1 ext = 1000; and IP phone2 ext = 1001.

I have a SIP provider that provides my SIP trunk with a DID number (5713132222).

I can make call successfull calls from IP phone1 to IPhone2 and vice versa.

I can make outgoing call over the SIP trunk to PSTN;

I can make incoming calls into my CME router via SIP trunk DID number.

ISSUE: i have no dial-peers configured for incoming calls via DID number into my CME router, but

when i make an incoming call via SIP trunk via DID number, the IP phone1 rings; let me say when making incoming DID call i'm dialing DID number (5713132222), but my IP phone1 with ext 1000 rings; I can answer and get talkpath.

How dailing the DID number makes my IP phone ext 1000 rings? When is the dial-peer that is converting from DID # to my ext #?

thanks

[heathrw]

Hi,

So you only have a single number or a range of numbers? 

Can you please attach the following files.

• A copy of your running config
• Run 'debug ccsip messages' make a call to your DID number from an external phone and attach the output.

[adsuther1]

Yes, I have a single number for DID.

[heathrw]

Are you going attach the other info?
 

[adsuther1]

I sent an attached file this morning;

But here is the Running-config and debug file

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.01.08 09:57:29 =~=~=~=~=~=~=~=~=~=~=~=
no debug alldebug ccsip messagesshow runn           
Building configuration...

Current configuration : 6659 bytes
!
! No configuration change since last restart
!
version 15.0
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname LAP_3825_1PORT
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$X8TI$9GWQ1D3/.FKSn5P4t0JXV0
!
no aaa new-model
!
!
!
clock timezone est -5
clock summer-time est recurring
no network-clock-participate wic 2
!
dot11 syslog
ip source-route
ip cef
!
!
ip dhcp excluded-address 192.168.x.x 192.168.x.x
ip dhcp excluded-address 192.168.x.x 192.168.x.x
ip dhcp excluded-address 192.168.x.x 192.168.x.x
ip dhcp excluded-address 192.168.x.x 192.168.x.x
!
ip dhcp pool DATA_NET
   network 192.168.x.x 255.255.255.0
   default-router 192.168.x.x
   dns-server 8.8.8.8
   lease 0 8
!
ip dhcp pool VOICE_NET
   network 192.168.x.x 255.255.255.0
   default-router 192.168.x.x
   option 150 ip 192.168.x.x
 --More--            dns-server 8.8.8.8
   lease 0 8
!
!
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
voice-card 0
!
!
!
voice service voip
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 sip
  registrar server expires max 3600 min 3600
  localhost dns:sip.nextiva.com
!
voice class codec 1
 codec preference 1 g711ulaw
!
!
!
!
!
crypto pki trustpoint TP-self-signed-1795450327
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-1795450327
 revocation-check none
 rsakeypair TP-self-signed-1795450327
!
!
crypto pki certificate chain TP-self-signed-1795450327
 certificate self-signed 01
  30820246 308201AF A0030201 02020101 300D0609 2A864886 F70D0101 04050030
  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
  69666963 6174652D 31373935 34353033 3237301E 170D3135 30313038 31353234
  32315A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
  4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 37393534
 --More--           35303332 3730819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
  8100E4EF 5E4F6C70 5BBE7294 E380E233 3AA138AD F7FE3F02 620C3647 3F07E2D9
  15BA8FF7 44B3D3DE 2545C73E 4E75459A B009D059 5DA7D046 00E15C0A 809E566B
  2503030F B132061E 91EF4700 C8B37EF8 77A28DBD 4902A31E 89267D34 C8991BAB
  21925BD9 3EB00675 DF575B4A 09798382 BC79727B 0C142BBF 2CD06CF8 12BC3A37
  0A890203 010001A3 6E306C30 0F060355 1D130101 FF040530 030101FF 30190603
  551D1104 12301082 0E4C4150 5F333832 355F3150 4F525430 1F060355 1D230418
  30168014 C841C642 4FD9B2AD 93AC67B5 7E6B8377 684A89DC 301D0603 551D0E04
  160414C8 41C6424F D9B2AD93 AC67B57E 6B837768 4A89DC30 0D06092A 864886F7
  0D010104 05000381 81003767 467B64AC 47557C30 5011FFD4 892170DF 0A86E871
  67C37A9A 2EB2D5DF BB53857D 1748459B C81B9202 F9E30468 D19A6261 C0D0473F
  8E41A72B 9CE04D81 C41CB132 23ED61F3 14B17D47 B1F42B6A 48623787 CBEAF4B6
  D1016A8D 067BDF44 978E52EF 876A6028 93F902CD A1F0B2B6 D8590287 C0B06A1A
  5DE021C4 B2D24A87 9C4C
      quit
!
!
license udi pid CISCO3825 sn FTX1112A2NV
username Anthony1 password 0 cisco
!
redundancy
!
!
controller T1 0/2/0
!
controller T1 0/2/1
!
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0
 no ip address
 duplex auto
 speed auto
 media-type rj45
 !
!
interface GigabitEthernet0/0.1
 description OUTSIDE INTERFACE
 encapsulation dot1Q 1 native
 --More--          ip address dhcp
 ip nat outside
 ip virtual-reassembly
!
interface GigabitEthernet0/0.2
 description VOICE VLAN INTERFACE
 encapsulation dot1Q 2
 ip address 192.168.x.x 255.255.255.0
 ip nat inside
 ip virtual-reassembly
!
interface GigabitEthernet0/0.3
 description DATA VLAN INTERFACE
 encapsulation dot1Q 3
 ip address 192.168.x.x 255.255.255.0
 ip nat inside
 ip virtual-reassembly
!
interface Service-Engine0/0
 ip unnumbered GigabitEthernet0/0.2
 service-module ip address 192.168.x.x 255.255.255.0
 service-module ip default-gateway 192.168.x.x
 !
!
interface GigabitEthernet0/1
 no ip address
 ip nat outside
 ip virtual-reassembly
 shutdown
 duplex auto
 speed auto
 media-type rj45
 !
!
ip forward-protocol nd
ip http server
ip http secure-server
!
!
ip nat inside source list 10 interface GigabitEthernet0/0.1 overload
ip route 192.168.x.x 255.255.255.255 Service-Engine0/0
!
access-list 10 permit 192.168.x.x 0.0.0.255
!
!
 --More--         !
!
!
tftp-server flash:phone/7940-7960/P00308000500.bin alias P00308000500.bin
tftp-server flash:phone/7940-7960/P00308000500.loads alias P00308000500.loads
tftp-server flash:phone/7940-7960/P00308000500.sb2 alias P00308000500.sb2
tftp-server flash:phone/7940-7960/P00308000500.sbn alias P00308000500.sbn
!
control-plane
 !
!
!
voice-port 0/0/0
!
voice-port 0/0/1
!
!
!
!
dial-peer voice 300 voip
 description *** outbound from sip trunk ***
 destination-pattern 1..........
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad
!
!
sip-ua
 authentication username 101146724 password 7 025F5C095E5F5D79151F
 no remote-party-id
 retry invite 2
 retry register 10
 retry options 3
 timers connect 100
 registrar dns:trunking.voipdnsservers.com expires 3600
 sip-server dns:trunking.voipdnsservers.com
 host-registrar
!
!
!
telephony-service
 authentication credential Admin cisco
 max-ephones 8
 --More--          max-dn 99
 ip source-address 192.168.2.1 port 2000
 url services http://192.168.x.x/voiceview/common/login.do
 url authentication http://192.168.x.x/CCMCIP/authenticate.asp  
 voicemail 2000
 max-conferences 12 gain -6
 web admin system name Admin password cisco
 dn-webedit
 time-webedit
 transfer-system full-consult
 create cnf-files version-stamp 7960 Jan 04 2015 12:01:16
!
!
ephone-dn  1  dual-line
 number 1000
 label "Anthony Sutherland"
 description Anthony's extension
 name Anthony Sutherland
 call-forward busy 2000
 call-forward noan 2000 timeout 18
 hold-alert 30 originator
!
!
ephone-dn  2  dual-line
 number 1001
 label "Vickie Sutherland"
 description Vickie's extension
 name Vickie Sutherland
 call-forward busy 2000
 call-forward noan 2000 timeout 18
 hold-alert 30 originator
!
!
ephone  1
 device-security-mode none
 description "Anthony Sutherland's 1000 phone"
 mac-address 001D.4596.17F6
 username "Anthony" password cisco
 type 7960
 button  1:1
!
!
!
ephone  2
 device-security-mode none
 --More--          description "Vickie Sutherland's 2000 phone"
 mac-address 001F.CA36.11E9
 username "Vickie" password cisco
 speed-dial 1 1000 label "Anthony"
 type 7960
 button  1:2
!
!
!
line con 0
line aux 0
line 194
 no activation-character
 no exec
 transport preferred none
 transport input all
 transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
 exec-timeout 0 0
 password ciscopress
 login
!
scheduler allocate 20000 1000
ntp server 184.73.235.44
end

LAP_3825_1PORT#
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LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#show runnno debug alldebug ccsip messages
SIP Call messages tracing is enabled
LAP_3825_1PORT#
Jan  8 14:57:57.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:trunking.voipdnsservers.com:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6E774

From: <sip:1000@trunking.voipdnsservers.com>;tag=30A5A0-2527

To: <sip:1000@trunking.voipdnsservers.com>

Date: Thu, 08 Jan 2015 14:57:57 GMT

Call-ID: 2F980FD8-968111E4-8002CB16-98DB8E00

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1420729077

CSeq: 53 REGISTER

Contact: <sip:1000@192.168.1.100:5060>

Expires:  3600

Supported: path

Content-Length: 0

Jan  8 14:57:57.747: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;received=68.100.31.170;branch=z9hG4bK6E774;rport=62265

From: <sip:1000@trunking.voipdnsservers.com>;tag=30A5A0-2527

To: <sip:1000@trunking.voipdnsservers.com>;tag=aprqrttbei2-6gkqk110000a3

Call-ID: 2F980FD8-968111E4-8002CB16-98DB8E00

Timestamp: 1420729077

CSeq: 53 REGISTER

Contact: <sip:1000@192.168.1.100:5060>;expires=45

Jan  8 14:57:57.867: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:1001@192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK6aqbou3010mhagp8a6k0.1

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed

Call-Id: pcst1420729078307423717210@192.168.201.117

Contact: <sip:5715357972@208.73.146.93:5060;transport=udp>

Content-Disposition: session; handling=required

Content-Length: 237

Content-Type: application/sdp

CSeq: 1 INVITE

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

In-Reply-To: 1006937118_12049738@207.155.147.62

Session-Expires: 1800;refresher=uas

Supported: timer

To: <sip:5713133270@192.168.101.100>

Max-Forwards: 70

v=0

o=Sonus_UAC 23050 8631 IN IP4 208.73.146.93

s=SIP Media Capabilities

c=IN IP4 208.73.146.93

t=0 0

m=audio 20234 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=maxptime:20

Jan  8 14:57:57.879: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK6aqbou3010mhagp8a6k0.1

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

To: <sip:5713133270@192.168.101.100>

Date: Thu, 08 Jan 2015 14:57:57 GMT

Call-ID: pcst1420729078307423717210@192.168.201.117

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Jan  8 14:57:57.879: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK6aqbou3010mhagp8a6k0.1

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

To: <sip:5713133270@192.168.101.100>;tag=30A668-C57

Date: Thu, 08 Jan 2015 14:57:57 GMT

Call-ID: pcst1420729078307423717210@192.168.201.117

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <sip:1001@192.168.1.100:5060>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Jan  8 14:58:06.975: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK6aqbou3010mhagp8a6k0.1

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

To: <sip:5713133270@192.168.101.100>;tag=30A668-C57

Date: Thu, 08 Jan 2015 14:57:57 GMT

Call-ID: pcst1420729078307423717210@192.168.201.117

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <sip:1001@192.168.1.100:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Session-Expires:  1800;refresher=uas

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 235

v=0

o=CiscoSystemsSIP-GW-UserAgent 0 2338 IN IP4 192.168.1.100

s=SIP Call

c=IN IP4 192.168.1.100

t=0 0

m=audio 17924 RTP/AVP 0 101

c=IN IP4 192.168.1.100

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

Jan  8 14:58:07.075: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK6aqbou3010mhagp8a6k0.1

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

To: <sip:5713133270@192.168.101.100>;tag=30A668-C57

Date: Thu, 08 Jan 2015 14:57:57 GMT

Call-ID: pcst1420729078307423717210@192.168.201.117

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <sip:1001@192.168.1.100:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-12.x

Session-Expires:  1800;refresher=uas

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 235

v=0

o=CiscoSystemsSIP-GW-UserAgent 0 2338 IN IP4 192.168.1.100

s=SIP Call

c=IN IP4 192.168.1.100

t=0 0

m=audio 17924 RTP/AVP 0 101

c=IN IP4 192.168.1.100

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

Jan  8 14:58:07.087: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:1001@192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKsl4880205oegdjh306d1.1

Call-Id: pcst1420729078307423717210@192.168.201.117

Content-Length: 0

CSeq: 1 ACK

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

To: <sip:5713133270@192.168.101.100>;tag=30A668-C57

Max-Forwards: 70

Jan  8 14:58:07.171: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:1001@192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKsl4880205oegdjh306d1.1

Call-Id: pcst1420729078307423717210@192.168.201.117

Content-Length: 0

CSeq: 1 ACK

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

To: <sip:5713133270@192.168.101.100>;tag=30A668-C57

Max-Forwards: 70

Jan  8 14:58:13.815: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:5715357972@208.73.146.93:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6F1E8C

From: <sip:5713133270@192.168.101.100>;tag=30A668-C57

To: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

Date: Thu, 08 Jan 2015 14:58:07 GMT

Call-ID: pcst1420729078307423717210@192.168.201.117

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1420729093

CSeq: 101 BYE

Reason: Q.850;cause=16

Content-Length: 0

Jan  8 14:58:13.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;received=68.100.31.170;branch=z9hG4bK6F1E8C;rport=62265

From: <sip:5713133270@192.168.101.100>;tag=30A668-C57

To: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.117:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK04204cc2

Call-ID: pcst1420729078307423717210@192.168.201.117

Timestamp: 1420729093

CSeq: 101 BYE

Content-Length: 0

Jan  8 14:58:16.071: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:trunking.voipdnsservers.com:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK7016E0

From: <sip:1001@trunking.voipdnsservers.com>;tag=30ED78-ECB

To: <sip:1001@trunking.voipdnsservers.com>

Date: Thu, 08 Jan 2015 14:58:16 GMT

Call-ID: 2FFCC3E8-968111E4-8003CB16-98DB8E00

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1420729096

CSeq: 53 REGISTER

Contact: <sip:1001@192.168.1.100:5060>

Expires:  3600

Supported: path

Content-Length: 0

Jan  8 14:58:16.135: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;received=68.100.31.170;branch=z9hG4bK7016E0;rport=62265

From: <sip:1001@trunking.voipdnsservers.com>;tag=30ED78-ECB

To: <sip:1001@trunking.voipdnsservers.com>;tag=aprqrttbei2-jihc4710000a3

Call-ID: 2FFCC3E8-968111E4-8003CB16-98DB8E00

Timestamp: 1420729096

CSeq: 53 REGISTER

Contact: <sip:1001@192.168.1.100:5060>;expires=46

LAP_3825_1PORT#debug ccsip messagesshow runn           no debug all
All possible debugging has been turned off
LAP_3825_1PORT#

[adsuther1]

Heathrw,

did you get a chance to look at the RUNNING-CONF?

cheers,

Anthony

[heathrw]

Hi Anthony,

The running config does not have any translations but the debugging is interesting. Not a setup I have seen before. and only had a brief chance to check it out.

It appears that your phone extensions are registering with your trunking provider as per the following from the debug:

REGISTER sip:trunking.voipdnsservers.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK6E774
From: <sip:1000@trunking.voipdnsservers.com>;tag=30A5A0-2527
To: <sip:1000@trunking.voipdnsservers.com>

What are numbers 5715357972 and 5713133270? The single number you mentioned that you have does not appear in any of the debugs.

The debug appears that 5715357972  rang 5713133270 and maybe the call landed on extension 1001, are you able to confirm any of this?

[adsuther1]

What are numbers 5715357972 and 5713133270? The single number you mentioned that you have does not appear in any of the debugs.

The 5715357972 = my cell phone;

The 5713133270 = DID # for SIP Trunk.

The debug appears that 5715357972  rang 5713133270 and maybe the call landed on extension 1001, are you able to confirm any of this?

Yes my ISSUE: when I dial DID # 5713133270, how does it know to terminate to extension 1000? If 1000 is not active it terminates to my other

extension 1001; The SIP Provider has to be doing some translations somewhere. Like you say there is no translation in my Running-config.

Any ideas???

[heathrw]

Attach the output of
show sip-ua register status

Turn on 'debug ccsip messasges' add the below config

sip-ua 
credentials username 101146724 password 7 025F5C095E5F5D79151F realm trunking.voipdnsservers.com
authentication username 101146724 password 7 025F5C095E5F5D79151F

Attach the output as possible changes
show sip-ua register status

make another call into your system and attach the ccsip debug output

[adsuther1]

sorry for the delay....

Here is what i did:

BEFOR CHANGE:

LAP_3825_1PORT#show sip-ua register status
Line          peer           expires(sec)  registered   P-Associated-URI
============  =============  ============  ===========  ================
1000                              20001       53            yes
1001                              20002       11            yes
LAP_3825_1PORT#
LAP_3825_1PORT#

ADDING:

sip-ua
credentials username 101146724 password 7 025F5C095E5F5D79151F realm trunking.voipdnsservers.com
authentication username 101146724 password 7 025F5C095E5F5D79151F

Make call incoming call into CME:

RESULTS: call not complete; MSG: the wireless customer you are calling is not available please try again later.

[heathrw]

Hi,

Can you make a call from your Cisco phone to external

What is status after the change?

What is the output of the 'debug ccsip messages' when making a call in after the change?

[adsuther1]

Can you make a call from your Cisco phone to external? Yes

can not make incoming call into CME.

What is status after the change?

can originate calls from CME outgoing calls to external.

can not receive incoming calls into CME.

What is the output of the 'debug ccsip messages' when making a call in after the change?

HERE GO: failed incoming call to CME

LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#show runndialplan number 991111 en                      config tu all   debug ccsip messagesgesterminal monitor     
LAP_3825_1PORT#terminal monitorshow runn       dialplan number 991111 en                      config tu all   debug ccsip message
SIP Call messages tracing is enabled
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
Jan 16 02:28:57.393: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:trunking.voipdnsservers.com:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK3238F

From: <sip:101146724@trunking.voipdnsservers.com>;tag=12E310-42B

To: <sip:101146724@trunking.voipdnsservers.com>

Date: Fri, 16 Jan 2015 02:28:57 GMT

Call-ID: D65DCF1C-9C5D11E4-8022CA0F-335C7C00

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1421375337

CSeq: 6 REGISTER

Contact: <sip:101146724@192.168.1.100:5060>

Expires:  3600

Supported: path

Content-Length: 0

Jan 16 02:28:57.473: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#
LAP_3825_1PORT#SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;received=68.100.31.170;branch=z9hG4bK3238F;rport=6216

From: <sip:101146724@trunking.voipdnsservers.com>;tag=12E310-42B

To: <sip:101146724@trunking.voipdnsservers.com>;tag=aprqrttbei2-ipb64a10000c0

Call-ID: D65DCF1C-9C5D11E4-8022CA0F-335C7C00

Timestamp: 1421375337

CSeq: 6 REGISTER

Contact: <sip:101146724@192.168.1.100:5060>;expires=46

Jan 16 02:29:04.169: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:101146724@192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKn2384l100ofhmp1ap6p1.1

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed

Call-Id: pcst14213753429072888275310@192.168.201.118

Contact: <sip:5715357972@208.73.146.93:5060;transport=udp>

Content-Disposition: session; handling=required

Content-Length: 238

Content-Type: application/sdp

CSeq: 1 INVITE

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.118:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK0e5afe94

In-Reply-To: 1007554466_48735224@207.155.147.62

Session-Expires: 1800;refresher=uas

Supported: timer

To: <sip:5713133270@192.168.101.100>

Max-Forwards: 70

v=0

o=Sonus_UAC 30485 20445 IN IP4 208.73.146.93

s=SIP Media Capabilities

c=IN IP4 208.73.146.93

t=0 0

m=audio 32714 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=maxptime:20

Jan 16 02:29:04.177: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKn2384l100ofhmp1ap6p1.1

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.118:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK0e5afe94

To: <sip:5713133270@192.168.101.100>

Date: Fri, 16 Jan 2015 02:29:04 GMT

Call-ID: pcst14213753429072888275310@192.168.201.118

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Jan 16 02:29:04.177: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKn2384l100ofhmp1ap6p1.1

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.118:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK0e5afe94

To: <sip:5713133270@192.168.101.100>;tag=12FD8C-122A

Date: Fri, 16 Jan 2015 02:29:04 GMT

Call-ID: pcst14213753429072888275310@192.168.201.118

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

Jan 16 02:29:04.261: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:101146724@192.168.1.100:5060 SIP/2.0

Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKn2384l100ofhmp1ap6p1.1

CSeq: 1 ACK

Call-ID: pcst14213753429072888275310@192.168.201.118

From: "AVAYA GOVERNMEN" <sip:5715357972@192.168.101.118:5060;otg=TG_CORE_GSX;pstn-params=808482808882>;tag=gK0e5afe94

To: <sip:5713133270@192.168.101.100>;tag=12FD8C-122A

Max-Forwards: 70

Content-Length: 0

LAP_3825_1PORT#debug ccsip messageterminal monitor   show runn       
Jan 16 02:29:42.838: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:trunking.voipdnsservers.com:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK33FED

From: <sip:1000@trunking.voipdnsservers.com>;tag=139494-218B

To: <sip:1000@trunking.voipdnsservers.com>

Date: Fri, 16 Jan 2015 02:29:42 GMT

Call-ID: 6CA3F7B9-9C6611E4-8002CA0F-335C7C00

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1421375382

CSeq: 20 REGISTER

Contact: <sip:1000@192.168.1.100:5060>

Expires:  3600

Supported: path

Content-Length: 0

Jan 16 02:29:42.918: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
  SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;received=68.100.31.170;branch=z9hG4bK33FED;rport=6216

From: <sip:1000@trunking.voipdnsservers.com>;tag=139494-218B

To: <sip:1000@trunking.voipdnsservers.com>;tag=aprqrttbei2-n7qi461000081

Call-ID: 6CA3F7B9-9C6611E4-8002CA0F-335C7C00

Timestamp: 1421375382

CSeq: 20 REGISTER

Contact: <sip:1000@192.168.1.100:5060>;expires=46

% Ambiguous command:  "show ru"
LAP_3825_1PORT#u
Jan 16 02:29:46.130: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:trunking.voipdnsservers.com:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK3489B

From: <sip:1001@trunking.voipdnsservers.com>;tag=13A170-0

To: <sip:1001@trunking.voipdnsservers.com>

Date: Fri, 16 Jan 2015 02:29:46 GMT

Call-ID: 6D09E418-9C6611E4-8003CA0F-335C7C00

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1421375386

CSeq: 23 REGISTER

Contact: <sip:1001@192.168.1.100:5060>

Expires:  3600

Supported: path

Content-Length: 0

Jan 16 02:29:46.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
 allSIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.100:5060;received=68.100.31.170;branch=z9hG4bK3489B;rport=6216

From: <sip:1001@trunking.voipdnsservers.com>;tag=13A170-0

To: <sip:1001@trunking.voipdnsservers.com>;tag=aprqrttbei2-ufeg3c00000e1

Call-ID: 6D09E418-9C6611E4-8003CA0F-335C7C00

Timestamp: 1421375386

CSeq: 23 REGISTER

Contact: <sip:1001@192.168.1.100:5060>;expires=46

All possible debugging has been turned off
LAP_3825_1PORT#

• Correct Answer 

[heathrw]

So if you now enter 'num-exp 5713133270 1001' make a call to your system it should ring on 1001. And if 1001 is not connected the call should fail or go to voice mail.

If you are happy with that you may want to add a secondary line to each phone so it rings on all phones at once, see below.

no num-exp 5713133270 1001
num-exp 5713133270  1100
!
ephone-dn  3  octo
 number 1100
 label "Main Line"
 description Main Line
 name Main Line
 call-forward busy 2000
 call-forward noan 2000 timeout 18
 hold-alert 30 originator
!
 ephone  1
  button 1:1 2:3
  restart
 !
 ephone  2
  button 1:3 2:3
  restart

let me know how you go