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Inbound SIP Calls... lost in the CUCME.

darksilicon
Level 1
Level 1

I am just having a devil of a time trying to get an inbound call from the sip trunk, which makes outbound calls just fine... to send an incoming call to an extension ... :/ this was easier with the phone cards.

Any ideas...    I'd just love to have all incoming calls goto extension 2000, etc...

!

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service h450.2

no supplementary-service h450.3

no supplementary-service h450.7

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

no supplementary-service sip handle-replaces

sip

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

!

!

!

voice translation-rule 1

rule 1 /5555555555/ /601/

!

!

voice translation-profile INBOUND

translate called 1

!

!

voice-card 0

!

!

!

!

!

!

!

!

!

interface FastEthernet0/0

description Connection to Internal Network

ip address 192.168.1.1 255.255.255.0

ip nbar protocol-discovery

ip nat inside

ip virtual-reassembly in

duplex auto

speed auto

!

interface FastEthernet0/1

description Connection to Broadstar Cable Modem

bandwidth 1435

bandwidth inherit 1433

bandwidth receive 12288

bandwidth qos-reference 1433

ip address dhcp client-id FastEthernet0/1

ip nbar protocol-discovery

ip nat outside

ip virtual-reassembly in

speed auto

full-duplex

service-policy output CCP-QoS-Policy-2

!

interface Serial0/1/0

no ip address

shutdown

!

ip forward-protocol nd

ip http server

ip http authentication local

ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

ip http path flash:/GUI

!

ip flow-top-talkers

top 5

sort-by bytes

cache-timeout 300000

!

ip dns server

ip nat inside source list 1 interface FastEthernet0/1 overload

!

access-list 1 permit 192.168.1.0 0.0.0.255

!

!

!

dial-peer voice 400 voip

destination-pattern 1[2-9].........

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 2 voip

!

dial-peer voice 1 voip

incoming called-number .

!

!

presence

!

sip-ua

XXXXX

nat symmetric role passive

no remote-party-id

retry invite 2

registrar dns:atl.teliax.net:5060 expires 360

sip-server dns:atl.teliax.net

!

!

!

telephony-service

no auto-reg-ephone

max-ephones 25

max-dn 25

ip source-address 192.168.1.1 port 2000

max-redirect 5

system message Cute White Cloud

cnf-file location flash:

load 7911 SCCP11.9-3-1-1S.loads

time-zone 12

max-conferences 4 gain -6

XXXXX

transfer-system full-consult

XXXXXX

create cnf-files version-stamp 7960 Sep 18 2012 03:44:27

!

!

ephone-dn  1  dual-line

number 2000

label Darek Bridges

description Darek's Desk Phone

name Darek Bridges

!

!

ephone-dn  2  dual-line

number 2001

label Raymond Goad

description Raymond Goad's Desk Phone

name Raymond Goad

!

!

ephone-dn  3  dual-line

number 2002

label Cody Jackson

description Cody Jackson's Desk Phone

name Cody Jackson

!

!

ephone-dn  4

number 5555555555

!

!

ephone  1

device-security-mode none

mac-address 0024.C444.A214

XXXXX

type 7911

!

!

!

ephone  2

device-security-mode none

mac-address 0024.C444.A373

XXXX

type 7911

button  1:2

!

!

!

ephone  3

device-security-mode none

XXX

type 7911

button  1:3

!

!

!

ephone  4

device-security-mode none

mac-address 0024.C444.AC87

XXX

type 7911

!

!

!

ephone  5

device-security-mode none

mac-address 0024.C444.A16F

XXX

type 7911

!

!

!

ephone  6

device-security-mode none

mac-address 001F.CAE9.E4D8

XXXX

type 7911

button  1:1

pin 0418

!

!

!

banner login ^CThis is a super neat router.^C

!

line con 0

line aux 0

line vty 0 4

privilege level 15

login local

transport input ssh

transport output ssh

line vty 5 15

privilege level 15

login local

transport input ssh

transport output ssh

!

scheduler allocate 20000 1000

ntp update-calendar

ntp server 128.227.205.3 source FastEthernet0/1

ntp server time.apple.com prefer

ntp server time.microsoft.com

end

29 Replies 29

Godzilla#show ephone registered

ephone-6[5] Mac:001F.CAE9.E4D8 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 22/17 max_streams=5

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9

IP:192.168.1.61 * 51678 7911  keepalive 3262 max_line 1 available_line 1

button 1: cw:1 ccw:(0 0)

  dn 1  number 2000 CH1   IDLE         CH2   IDLE        

Preferred Codec: g711ulaw

Lpcor Type: none Username: user3 Password: hippo

Godzilla#

It seams that you only have extension 2000 registered and your latest test calls has been to 2002.

dn 1 number 2000 CH1 IDLE

Could you try to make a test call to extension 2000. If that isn't successful please post the output of the same debug that you previously have captured.

Please rate useful posts.

Sent from Cisco Technical Support iPhone App



Response Signature


Ok. You dont have any phone with extension 2002. All I see is a phone with extension 2000.

Please configure this translation profile and apply it to your dial-peer 1 as follows:

conf t

voice translation-rule 10

rule 1 /^200./ /2000/

voice translation-profile DID

translate called 10

dial-peer voice 1 voip

translation-profile incoming DID

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

Okay so extension 2000 rings when you call the number... can you explain a bit what these commands did exactly... i understand digit translation... etc...

==============================================

MiniMoos:~ Braden$ ssh darek@192.168.1.1

This is a super neat router.Password:

Godzilla#conf t

Enter configuration commands, one per line.  End with CNTL/Z.

Godzilla(config)#

Godzilla(config)#voice translation rule 10

                                        ^

% Invalid input detected at '^' marker.

Godzilla(config)#voice translation rule 10

                                        ^

% Invalid input detected at '^' marker.

Godzilla(config)#voice translation-rule 10

Godzilla(cfg-translation-rule)#rule 1 /^200./ /2000/

Godzilla(cfg-translation-rule)#voice translation-profile DID

Godzilla(cfg-translation-profile)#translate called 10

Godzilla(cfg-translation-profile)#dial-peer voice 1 voip

Godzilla(config-dial-peer)#translation-profile incoming DID

Godzilla(config-dial-peer)#exit

Godzilla(config)#exit

Godzilla#

voice translation-rule 10

! Creates a voice translation rule with tag 10

rule 1 /^200./ /2000/

! Rule 1 in rule set 10.

Left part is the match and right translation part. This rule matches on start with, that's the ^, 200, and then any digit, the . This is translated to 2000

voice translation-profile DID

! Creates a translation profile with name DID

translate called 10

! Tie rule set 10 to translate called number

dial-peer voice 1 voip

! Go into config of dial peer 1

translation-profile incoming DID

! Tie translation profile DID in the incoming direction.

In short, all incoming calls that start with 200 and any more digit will be translated to 2000.

So all calls will go to that extension, if you don't want that all call goes to one phone you need to remove the translation and connect more phones so that they can register.

Please rate useful posts.

Sent from Cisco Technical Support iPhone App



Response Signature


Well I managed to screw this up again... so i added a SIP Phone (iPhone Jabber) on extension 3000... (are you supposed to use 2000, 3000, etc.. as extension numbers? )

starting then..

all new inbound calls were being routed to the 3000 number... so i deleted that sip phone and its extension. and thought the calls would start coming into the 2000 number but no dice... now when people call the CME i get a disconnected message...

calling out still works fine.

Perhaps I need to understand at a more generic level how inbound SIP trunk calls are routed into the CUCME.. :/

Godzilla#

Godzilla#

Godzilla#

Godzilla#

Godzilla#

Godzilla#

Godzilla#

Godzilla#

Godzilla#config t

Enter configuration commands, one per line.  End with CNTL/Z.

Godzilla(config)#dial-peer

% Incomplete command.

Godzilla(config)#

Godzilla(config)#

Godzilla(config)#

Godzilla(config)#Connection to 192.168.1.1 closed by remote host.

Connection to 192.168.1.1 closed.

MiniMoos:~ Braden$ ssh darek@192.168.1.1

This is a super neat router.Password:

Godzilla#show ephone registered

ephone-6[5] Mac:001F.CAE9.E4D8 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 22/17 max_streams=5

mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9

IP:192.168.1.61 * 49154 7911  keepalive 1445 max_line 1 available_line 1

button 1: cw:1 ccw:(0 0)

  dn 1  number 2003 CH1   IDLE         CH2   IDLE        

Preferred Codec: g711ulaw

Lpcor Type: none Username: user3 Password: hippo

Godzilla#

Godzilla#

Godzilla#debug voip ccapi inout

voip ccapi inout debugging is on

Godzilla#show terminal monitor

                       ^

% Invalid input detected at '^' marker.

Godzilla#terminal monitor

Godzilla#

Sep 26 14:44:25.614: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:3000@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKg9SUHgNFNc6Sc

Max-Forwards: 70

From: ;tag=QN9NNcgtHHXtS

To: <>8664@atl.teliax.net>

Call-ID: 824bb243-828b-1230-85aa-001143d2f397

CSeq: 131564646 OPTIONS

Contact:

User-Agent: Teliaware2.0.2

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

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Length: 0

Sep 26 14:44:25.626: //29242/808614B8B26B/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKg9SUHgNFNc6Sc

From: ;tag=QN9NNcgtHHXtS

To: <>8664@atl.teliax.net>;tag=3B341940-170C

Date: Wed, 26 Sep 2012 14:44:25 GMT

Call-ID: 824bb243-828b-1230-85aa-001143d2f397

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 131564646 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 170

v=0

o=CiscoSystemsSIP-GW-UserAgent 5503 5873 IN IP4 12.202.42.166

s=SIP Call

c=IN IP4 12.202.42.166

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 12.202.42.166

Sep 26 14:44:31.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:5612002526@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKXZBQ1HBeKXg3N

Max-Forwards: 25

From: "WASHINGTON   DC" <2024558888>;tag=graUtjg3cDK4D

To: <5612002526>

Call-ID: 8603c579-828b-1230-85aa-001143d2f397

CSeq: 33999271 INVITE

Contact:

User-Agent: Teliaware2.0.2

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

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 223

x-caller-dnis: 5612002526

x-teliax-accountcode: 64605679

x-teliax-version: 2

x-user-status: active

x-international: false

X-FS-Support: update_display

Remote-Party-ID: "WASHINGTON   DC" <2024558888>;party=calling;screen=yes;privacy=off

v=0

o=FreeSWITCH 1348640296 1348640297 IN IP4 8.17.37.23

s=FreeSWITCH

c=IN IP4 8.17.37.23

t=0 0

m=audio 30376 RTP/AVP 0 9 18 101 13

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Sep 26 14:44:31.870: //-1/843E3B0CB26C/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=2024558888

   ----- ccCallInfo IE subfields -----

   cisco-ani=2024558888

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=5612002526

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=0

   cisco-rdnplan=0

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

Sep 26 14:44:31.870: //-1/843E3B0CB26C/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x697751E8, Call Info(

   Calling Number=2024558888,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=5612002526(TON=Unknown, NPI=Unknown),

   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,

   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=29243

Sep 26 14:44:31.870: //-1/843E3B0CB26C/CCAPI/ccCheckClipClir:

   In: Calling Number=2024558888(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Sep 26 14:44:31.870: //-1/843E3B0CB26C/CCAPI/ccCheckClipClir:

   Out: Calling Number=2024558888(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Sep 26 14:44:31.870: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Sep 26 14:44:31.870: :cc_get_feature_vsa malloc success

Sep 26 14:44:31.870: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Sep 26 14:44:31.870:  cc_get_feature_vsa count is 1

Sep 26 14:44:31.870: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Sep 26 14:44:31.870: :FEATURE_VSA attributes are: feature_name:0,feature_time:1794004640,feature_id:862

Sep 26 14:44:31.870: //29243/843E3B0CB26C/CCAPI/cc_api_call_setup_ind_common:

   Set Up Event Sent;

   Call Info(Calling Number=2024558888(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=5612002526(TON=Unknown, NPI=Unknown))

Sep 26 14:44:31.874: //29243/843E3B0CB26C/CCAPI/cc_process_call_setup_ind:

   Event=0x6A266768

Sep 26 14:44:31.874: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:

   Try with the demoted called number 5612002526

Sep 26 14:44:31.878: //29243/843E3B0CB26C/CCAPI/ccCallSetContext:

   Context=0x6BF6B078

Sep 26 14:44:31.878: //29243/843E3B0CB26C/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 29243 with tag 1 to app "_ManagedAppProcess_Default"

Sep 26 14:44:31.882: //29243/843E3B0CB26C/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKXZBQ1HBeKXg3N

From: "WASHINGTON   DC" <2024558888>;tag=graUtjg3cDK4D

To: <5612002526>

Date: Wed, 26 Sep 2012 14:44:31 GMT

Call-ID: 8603c579-828b-1230-85aa-001143d2f397

CSeq: 33999271 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Sep 26 14:44:31.882: //29243/843E3B0CB26C/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

Sep 26 14:44:31.886: //29243/843E3B0CB26C/CCAPI/ccCallDisconnect:

   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Sep 26 14:44:31.886: //29243/843E3B0CB26C/CCAPI/ccCallDisconnect:

   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)

Sep 26 14:44:31.890: //29243/843E3B0CB26C/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKXZBQ1HBeKXg3N

From: "WASHINGTON   DC" <2024558888>;tag=graUtjg3cDK4D

To: <5612002526>;tag=3B3431BC-7D4

Date: Wed, 26 Sep 2012 14:44:31 GMT

Call-ID: 8603c579-828b-1230-85aa-001143d2f397

CSeq: 33999271 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

Sep 26 14:44:31.926: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:5612002526@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKXZBQ1HBeKXg3N

Max-Forwards: 25

From: "WASHINGTON   DC" <2024558888>;tag=graUtjg3cDK4D

To: <5612002526>;tag=3B3431BC-7D4

Call-ID: 8603c579-828b-1230-85aa-001143d2f397

CSeq: 33999271 ACK

Content-Length: 0

Sep 26 14:44:31.930: //29243/843E3B0CB26C/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x697751E8, Tag=0x0, Call Id=29243,

   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)

Sep 26 14:44:31.930: //29243/843E3B0CB26C/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Sep 26 14:44:31.930: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Sep 26 14:44:31.930: :cc_free_feature_vsa freeing 6AEE5698

Sep 26 14:44:31.930: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Sep 26 14:44:31.930:  vsacount in free is 0

Sep 26 14:44:33.662: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:5612002526@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKy84F3cvHg66NH

Max-Forwards: 15

From: "WASHINGTON   DC" <2024558888>;tag=mvFy1yKH1gcFc

To: <5612002526>

Call-ID: 8717ae4a-828b-1230-85aa-001143d2f397

CSeq: 33999272 INVITE

Contact:

User-Agent: Teliaware2.0.2

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

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 223

x-caller-dnis: 5612002526

x-teliax-accountcode: 64605679

x-teliax-version: 2

x-user-status: active

x-international: false

X-FS-Support: update_display

Remote-Party-ID: "WASHINGTON   DC" <2024558888>;party=calling;screen=yes;privacy=off

v=0

o=FreeSWITCH 1348654036 1348654037 IN IP4 8.17.37.23

s=FreeSWITCH

c=IN IP4 8.17.37.23

t=0 0

m=audio 16638 RTP/AVP 0 9 18 101 13

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Sep 26 14:44:33.678: //-1/85521CFAB271/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=2024558888

   ----- ccCallInfo IE subfields -----

   cisco-ani=2024558888

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=5612002526

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=0

   cisco-rdnplan=0

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

Sep 26 14:44:33.678: //-1/85521CFAB271/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x697751E8, Call Info(

   Calling Number=2024558888,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=5612002526(TON=Unknown, NPI=Unknown),

   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,

   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=29244

Sep 26 14:44:33.678: //-1/85521CFAB271/CCAPI/ccCheckClipClir:

   In: Calling Number=2024558888(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Sep 26 14:44:33.678: //-1/85521CFAB271/CCAPI/ccCheckClipClir:

   Out: Calling Number=2024558888(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)

Sep 26 14:44:33.678: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Sep 26 14:44:33.678: :cc_get_feature_vsa malloc success

Sep 26 14:44:33.678: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Sep 26 14:44:33.678:  cc_get_feature_vsa count is 1

Sep 26 14:44:33.678: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Sep 26 14:44:33.678: :FEATURE_VSA attributes are: feature_name:0,feature_time:1794004640,feature_id:863

Sep 26 14:44:33.682: //29244/85521CFAB271/CCAPI/cc_api_call_setup_ind_common:

   Set Up Event Sent;

   Call Info(Calling Number=2024558888(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=5612002526(TON=Unknown, NPI=Unknown))

Sep 26 14:44:33.682: //29244/85521CFAB271/CCAPI/cc_process_call_setup_ind:

   Event=0x6A266768

Sep 26 14:44:33.682: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:

   Try with the demoted called number 5612002526

Sep 26 14:44:33.682: //29244/85521CFAB271/CCAPI/ccCallSetContext:

   Context=0x6BF6C538

Sep 26 14:44:33.686: //29244/85521CFAB271/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 29244 with tag 1 to app "_ManagedAppProcess_Default"

Sep 26 14:44:33.686: //29244/85521CFAB271/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

Sep 26 14:44:33.690: //29244/85521CFAB271/CCAPI/ccCallDisconnect:

   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Sep 26 14:44:33.690: //29244/85521CFAB271/CCAPI/ccCallDisconnect:

   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)

Sep 26 14:44:33.694: //29244/85521CFAB271/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKy84F3cvHg66NH

From: "WASHINGTON   DC" <2024558888>;tag=mvFy1yKH1gcFc

To: <5612002526>

Date: Wed, 26 Sep 2012 14:44:33 GMT

Call-ID: 8717ae4a-828b-1230-85aa-001143d2f397

CSeq: 33999272 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Sep 26 14:44:33.698: //29244/85521CFAB271/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKy84F3cvHg66NH

From: "WASHINGTON   DC" <2024558888>;tag=mvFy1yKH1gcFc

To: <5612002526>;tag=3B3438C8-1146

Date: Wed, 26 Sep 2012 14:44:33 GMT

Call-ID: 8717ae4a-828b-1230-85aa-001143d2f397

CSeq: 33999272 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

Sep 26 14:44:33.730: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:5612002526@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKy84F3cvHg66NH

Max-Forwards: 15

From: "WASHINGTON   DC" <2024558888>;tag=mvFy1yKH1gcFc

To: <5612002526>;tag=3B3438C8-1146

Call-ID: 8717ae4a-828b-1230-85aa-001143d2f397

CSeq: 33999272 ACK

Content-Length: 0

Sep 26 14:44:33.734: //29244/85521CFAB271/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x697751E8, Tag=0x0, Call Id=29244,

   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)

Sep 26 14:44:33.734: //29244/85521CFAB271/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Sep 26 14:44:33.734: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Sep 26 14:44:33.734: :cc_free_feature_vsa freeing 6AEE5698

Sep 26 14:44:33.734: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Sep 26 14:44:33.734:  vsacount in free is 0

Sep 26 14:44:56.222: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:3000@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKS0KjpeFmgB47m

Max-Forwards: 70

From: ;tag=7D6crBpSZ88Qr

To: <>8664@atl.teliax.net>

Call-ID: 9489e624-828b-1230-85aa-001143d2f397

CSeq: 131564835 OPTIONS

Contact:

User-Agent: Teliaware2.0.2

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

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Length: 0

Sep 26 14:44:56.234: //29245/92C3E8FAB276/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKS0KjpeFmgB47m

From: ;tag=7D6crBpSZ88Qr

To: <>8664@atl.teliax.net>;tag=3B3490CC-1D8E

Date: Wed, 26 Sep 2012 14:44:56 GMT

Call-ID: 9489e624-828b-1230-85aa-001143d2f397

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 131564835 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 170

v=0

o=CiscoSystemsSIP-GW-UserAgent 6841 1008 IN IP4 12.202.42.166

s=SIP Call

c=IN IP4 12.202.42.166

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 12.202.42.166

Sep 26 14:45:27.011: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:3000@12.202.42.166:57330 SIP/2.0

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKrXgH9X0N8g3tS

Max-Forwards: 70

From: ;tag=rg5NN77ce6KyN

To: <>8664@atl.teliax.net>

Call-ID: a6e32253-828b-1230-85aa-001143d2f397

CSeq: 131565024 OPTIONS

Contact:

User-Agent: Teliaware2.0.2

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

Supported: timer, precondition, path, replaces

Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer

Content-Length: 0

Sep 26 14:45:27.023: //29246/A51DD097B277/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 8.17.37.23;rport;branch=z9hG4bKrXgH9X0N8g3tS

From: ;tag=rg5NN77ce6KyN

To: <>8664@atl.teliax.net>;tag=3B350910-16B9

Date: Wed, 26 Sep 2012 14:45:27 GMT

Call-ID: a6e32253-828b-1230-85aa-001143d2f397

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 131565024 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 170

v=0

o=CiscoSystemsSIP-GW-UserAgent 9547 9231 IN IP4 12.202.42.166

s=SIP Call

c=IN IP4 12.202.42.166

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 12.202.42.166

So what is the problem Darek? What number are you dialling?

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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there is only one inbound number --  561-200-2526  -- right now it should be translating that number to 2003 to ring that extension...

I just thought I understood this conceptually .. and now it appears that I don't... Inbound calls are unable to reach 2003... (i was told that 2000 was a bad extension and I should use 2001, 2002, etc...)

configure this..

conf t

voice translation-rule 10

rule 1 /^5612002526 / /2003/

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Godzilla(cfg-translation-rule)#

Godzilla(cfg-translation-rule)#rule 1 /^5612002526 / /2003/

                                       ^

% Invalid input detected at '^' marker.

Godzilla(cfg-translation-rule)#rule 1 /^5612002526 /  /2003/

                                       ^

% Invalid input detected at '^' marker.

Godzilla(cfg-translation-rule)#

rule 1 /^5612002526/ /2003/ (note there is no space btw the last digit and /

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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well it took.. however phone still isnt ringing.

I assume that the other part of the config is still intact..

voice translation-profile DID

translate called 10

dial-peer voice 1 voip

translation-profile incoming DID

please send debugs

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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It sounds that you really need to start at the basic.

First check the call path with debug voip dialpeer. After you have established what DP that is used for incoming and outgoing you can take the next step.

Check your voice translation rule with debug voice translation and you can also test it directly with test voice translation-rule rulenumber numbertotest

See more about the test command at this URL. http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

Then when you know that your number translation are correct you need to ensure that your CME phone is registered and have the extension that you have translated to. Do that with show ephone reg.

Normally you would do a translation of the incoming DID to the internal extensions, for example like this.

DID range 561-200-25xx translate to internal extensions 25xx. The logic is to use part of the DID numbers as your internal numbers. How many digits that you use is based on whatever fits into your internal number plan.

Also please read some of the many configuration guides and examples that is available on cisco.com. For example search for configure call manager express.

I did a quick search and found this, http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_installation_and_configuration_guides_list.html

There are plenty more on both cisco.com and other places.

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