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Callcentric SIP Trunk (ITSP --> 2811 CUBE --> CUCM 8.6

whatuusay1
Level 1
Level 1

I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.

I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running  15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.

I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html

I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error.  I think I’m close.. but who knows. Any assistance would be greatly appreciated

voice class sip-profiles 1

request INVITE peer-header sip TO copy ".sip:(.*)@." u01

request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"

CUCM (single/pub)- 192.168.1.200

2811 acting as cube - 192.168.1.203

Calling Number - 18165297500

Called Number - 18452055544

vrtr1#show  sip register status

Line                             peer       expires(sec) registered P-Associ-URI

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

17772253754                      -1         20           yes

vrtr1#

The Call Setup Information is:

Call Control Block (CCB) : 0x49646C28

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 18165297500

Called Number            : 17772253754 (my customer number not called number)

Source IP Address (Sig  ): 192.168.1.203 (my 2811 router)

Destn SIP Req Addr:Port  : 204.11.192.159:5080

Destn SIP Resp Addr:Port : 204.11.192.159:5080

Destination Name         : 204.11.192.159

Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

f: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

t: <sip:18452055544@ss.callcentric.com>

i: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 INVITE

Max-Forwards: 8

m: <sip:ca7fe50c9cebe12327fe0d63c5962a3e@204.11.192.159:5080;transport=udp>

Supported: timer

c: application/sdp

l: 350

v=0

o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159

s=sip call

c=IN IP4 204.11.192.159

t=0 0

m=audio 61094 RTP/AVP 18 0 8 101

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=ptime:20

a=sendrecv

a=silenceSupp:off - - - -

a=setup:actpass

Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

From: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

To: <sip:18452055544@ss.callcentric.com>

Date: Fri, 14 Feb 2014 17:20:53 GMT

Call-ID: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

From: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

To: <sip:18452055544@ss.callcentric.com>;tag=35399D8-63

Date: Fri, 14 Feb 2014 17:20:53 GMT

Call-ID: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

f: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

t: <sip:18452055544@ss.callcentric.com>;tag=35399D8-63

i: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 ACK

Max-Forwards: 10

l: 0

u all

Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:18452055544;cic=0288;rn=6465471001;npdi@alpha14.callcentric.com:5070 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb

f: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

t: <sip:18452055544@ss.callcentric.com>;tag=35399D8-63

i: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 ACK

Max-Forwards: 8

l: 0

************************** Running Config **************************

sh run
vrtr1#sh running-config
Building configuration...


Current configuration : 4189 bytes
!
! Last configuration change at 00:34:03 CST Fri Feb 14 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
!
hostname vrtr1
!
boot-start-marker
boot system flash:
boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
boot-end-marker
!
!
card type t1 0 0
logging buffered 4096 notifications
enable password cisco
!
no aaa new-model
memory-size iomem 5
clock timezone CST -6 0
clock summer-time CST recurring
no network-clock-participate wic 0
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
!
ip name-server 192.168.1.9
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice service voip
ip address trusted list
  ipv4 192.168.1.0 255.255.255.0
  ipv4 204.11.192.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
  registrar server expires max 1800 min 1800
  localhost dns:callcentric.com
  outbound-proxy dns:callcentric.com
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
!
!
!
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO2811 sn FTX1133A4QR
!
!
controller T1 0/0/0
cablelength long 0db
!
!
!
!
!
interface FastEthernet0/0
description ** LAN **
ip address 192.168.1.203 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.1.203
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
snmp mib persist circuit
!
!
control-plane
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.1.200 
ccm-manager config
!
mgcp
mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
!
mgcp profile default
!
!
dial-peer voice 999100 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 999101 pots
service mgcpapp
port 0/1/1
!
dial-peer voice 999102 pots
service mgcpapp
port 0/1/2
!
dial-peer voice 999103 pots
service mgcpapp
port 0/1/3
!
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 6 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
!
!
sip-ua
credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
host-registrar
!
!
!
line con 0
line aux 0
line vty 0 4
password cisco
login
transport input all
!
scheduler allocate 20000 1000
ntp server 199.102.46.72
ntp server 23.227.162.123 prefer
end

exit

47 Replies 47

Aok - Not having any luck with inbound routing. I tried specifying the Sip Header as well as Peer-header To field regardless both seem to be grabbing the 17772253754 (username) and no the dialed number 18452055545.   I havent played with outbound yet - I assume that part should be fairly easy (ha..ha..).

I've attached ccsip all debugs when using the recommended profile options for an inbound call.

voice service voip

!

voice class sip-profiles 5

response 200 sip-header Require remove

request INVITE sip-header TO copy "<>" u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@192.168.1.200:5060 SIP/2.0"

!

012348: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_sip_copy_pattern: sed_match succeeded

012349: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_prefix_slash_in_copy_var_val: ret_dst: 17772253754

012350: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_application_sip_copy_pattern: SIP Profiles COPY variable: u1 val: 17772253754

012351: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Req URI before modification : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

012352: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: Node found: COPY variable: u1 val: 17772253754

012353: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: substituted_replace_pattern : INVITE sip:17772253754

012354: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: configured_replace_pattern : @192.168.1.200:5060 SIP/2.0

012355: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: Final substituted_replace_pattern : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

012356: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Passing substituted replace pattern

012357: Feb 18 22:40:27.107: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Req URI after modification : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

012358: Feb 18 22:40:27.107: //3655/C9B844B581F3/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately

012359: Feb 18 22:40:27.107: //3655/C9B844B581F3/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0

012360: Feb 18 22:40:27.107: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:192.168.1.200, rport:5060 with laddr:192.168.1.203012348: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_sip_copy_pattern: sed_match succeeded
012349: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_prefix_slash_in_copy_var_val: ret_dst: 17772253754
012350: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_application_sip_copy_pattern: SIP Profiles COPY variable: u1 val: 17772253754
012351: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Req URI before modification : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0
012352: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: Node found: COPY variable: u1 val: 17772253754
012353: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: substituted_replace_pattern : INVITE sip:17772253754
012354: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: configured_replace_pattern : @192.168.1.200:5060 SIP/2.0
012355: Feb 18 22:40:27.103: //3655/C9B844B581F3/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: Final substituted_replace_pattern : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0
012356: Feb 18 22:40:27.103: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Passing substituted replace pattern
012357: Feb 18 22:40:27.107: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Req URI after modification : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0
012358: Feb 18 22:40:27.107: //3655/C9B844B581F3/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
012359: Feb 18 22:40:27.107: //3655/C9B844B581F3/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
012360: Feb 18 22:40:27.107: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:192.168.1.200, rport:5060 with laddr:192.168.1.203

INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.169:5080;branch=z9hG4bK-716633dc62c1fef90469b53084713f17

f: <18165297500>;tag=3601772862-354895

t: <>18452055545@ss.callcentric.com>

voice class sip-profiles 5

request INVITE peer-header t copy "<>" u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@192.168.1.200:5060 SIP/2.0

If it doesnt work..please send only the ff debugs

debug ccsip info

debug ccsip messages

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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  Aok - I added the following sip profile. The context is a bit different (It didnt accept what you provided).

voice class sip-profiles 5

request INVITE peer-header sip t copy "<>" u01

response 200 sip-header Require remove

request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@192.168.1.200:5060 SIP/2.0"

Ok..That doesnt look good..

Can you try this..

no request INVITE peer-header sip t copy "<>" u01--delete this one..

request INVITE peer-header sip To copy "<>" u01---add this one..

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts

  Aok - I adjusted the profile statement as sugested, here's the current sip profile, and the debug (same error on inbound).

voice class sip-profiles 5

request INVITE peer-header sip To copy "<>" u01

response 200 sip-header Require remove

request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@192.168.1.200:5060 SIP/2.0"

016005: Feb 19 15:12:55.210: //4668/6F2900C6823E/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:@192.168.1.200:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK12E1198A

From: <>18165297500@callcentric.com>;tag=101C3448-EB1

To: <17772253754>

Date: Wed, 19 Feb 2014 21:12:55 GMT

Call-ID:

6F33FD5F-98E111E3-8244E4C3-DC5CB4B2@192.168.1.203

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

Min-SE:  1800

Cisco-Guid: 1864958150-2564887011-2185159875-3697063090

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

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

CSeq: 101 INVITE

Timestamp: 1392844375

Contact: <18165297500>

Expires: 180

Allow-Events: telephone-event

Okay,

I have spent hours tyring to figure this out and I have just finished testing this in my environmet...

First of all, please remove all the previous sip profiles for the To header copy and add only this one..

request INVITE sip-header To copy "<>" u01

request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "\u01@\1"

Okay thats the correct syntax...However we have a slight issue...

The To header that this profile acts on is the the one sent to CUCM. And that To header is this...

Sent:

INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK8EF06

From: <>18165297500@callcentric.com>;tag=3E2A54-19B0

To: <17772253754>

You can then this is the reason CUBE is storing the value of the variable u01 as 17772253754.

The To value from the original INVITE is lost completely on the INVITE sent to CUCM..That means we are stick!!

To form the To field sent out, CUBE copies the Request URI from the orginal INVITE and just appends its IP address..

This is tough...I have been on this since 12am and its 3.30am UK time now and I cant find a way around this other you opening a TAC case or ask your provider to send the called number in the Request-URI

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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Lets try this..

voice class sip-copylist 1

sip-header To

Then add the voice class copy list tot he inbound dial-peer from ITSP..

dial-peer voice 6 voip

voice class sip-copylist 1

Then test with the sip profile above...

Send me debug ccsip info and debug ccsip messages

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

Please rate all useful posts

Aok - I followed this example - which is  exactly what I'm trying to do, and where I beleive you were going with your last post. Re-write the Invite with information from the To field using the sip-copy list.

http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/

Copied everything verbatim - and it doesnt work. I can't explain any of it. No matter what I do it will not match the inbound dial-peer coming from Callcentric (which is required to perform the re-write of the header).

I'm about to give up and go be a farmer - seriously this can't be that hard. I've memorized the stupid dial-peer matching guide.

The router selects an inbound dial peer by matching the information elements in the setup message with the dial peer attributes. The router attempts to match these items in the following order:

1. Called number with incoming called-number

2. Calling number with answer-address

3. Calling number with destination-pattern

4. Incoming voice port with configured voice port

So this should work right? Since its matching on the Incoming Called Number.. the first in order for inbound dial-peer matching right? Nope..

dial-peer voice 99 voip

description incoming SIP Trunk

translation-profile incoming Incoming

session protocol sipv2

session target sip-server

incoming called-number 17772253754

voice-class codec 1 

dtmf-relay rtp-nte

no vad

Or maybe the dial-peer isnt 'active' right? Nope - its up and active.

apsc-vrtr1#show dial-peer voice summary
dial-peer hunt 0
             AD                                    PRE PASS                OUT
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE
99910- pots  up   up                                0                      down 0/1/0 0
99910- pots  up   up                                0                      down 0/1/1 1
99910- pots  up   up                                0                      down 0/1/2 2
99910- pots  up   up                                0                      down 0/1/3 3
99901- pots  up   up                                0                      down 0/1/0 0
99     voip  up   up                                0  syst sip-server
1      voip  up   up             18452055545        0  syst ipv4:192.168.1.200

Maybe the translations aren't right? Nope - those work to.

apsc-vrtr1#test voice translation-rule 9 17772253754
Matched with rule 1
Original number: 17772253754    Translated number: 18452055545
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none

So then it should match my outdoing dial peer to CUCM right?

dial-peer voice 1 voip
description to/from cucm publisher
destination-pattern 18452055545
session protocol sipv2
session target ipv4:192.168.1.200
voice-class codec 1 
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad

Nope.. doesnt work .. any of it. 

I 'love' parsing through the debug where it saying "Match not found on Incoming called number: 17772253754"  even those thats exactly on the inbound dial-peer.

Dont give up yet..I opened a TAC case today because of this issue!!  As we speak I am trying to simulate this and use the work around  provided by TAC.not too different from what we have done so far..

I will come back to you shortly..Are you available to work on this right now?? Do you have webex?

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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I'm happy to give you the login info for my callcentric account - its just my lab account - you are free to build the entire thing in your lab to replicate the issue if you'd like (send me an email to andrew [at] kcexec.com outside the forum here and I'll provide the login detail).

Yes - I have a webex/teamviewer access if you want to hop on and look at the issue that way.

Thanks,

Andrew

First of all remove the translation profile uder this dial-peer..

dial-peer voice 99 voip

description incoming SIP Trunk

no translation-profile incoming Incoming

2..change the destination-pattern on this dial-peer to17772253754

dial-peer voice 1 voip

description to/from cucm publisher

destination-pattern 17772253754

session protocol sipv2

session target ipv4:192.168.1.200

voice-class codec 1 

voice-class sip profiles 1

dtmf-relay rtp-nte

no vad

Make only this changes and send me the ff:

debug ccsip info

debug ccsip mess

debug voip ccapi inout

I am waiting!

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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Here's the debugs with the requested changes.

dial-peer voice 99 voip

description incoming SIP Trunk

session protocol sipv2

session target sip-server

incoming called-number 17772253754

voice-class codec 1

voice-class sip copy-list 1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 1 voip

description to/from cucm publisher

destination-pattern 17772253754

session protocol sipv2

session target ipv4:192.168.1.200

voice-class codec 1

voice-class sip profiles 1

dtmf-relay rtp-nte

no vad

....................... SO HAPPY... cant explain any of it... debug says no match yet it still matches and adjusts the header translation and sends the call out to CUCM with the correct modified header. I'll be interested to see what you think.

019021: Feb 20 15:01:11.729: //6176/F644081C837B/SIP/Info/sipSPISendInvite: Associated container=0x49A93EA4 to Invite
019022: Feb 20 15:01:11.733: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Req URI before modification : INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0
019023: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: Node found: COPY variable: u1 val: 18452055545
019024: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: substituted_replace_pattern : INVITE sip:18452055545
019025: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: configured_replace_pattern : @\1
019026: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: substituted_replace_pattern : INVITE sip:18452055545@\
019027: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: configured_replace_pattern : 1
019028: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sip_profiles_check_and_get_variables_in_replace_pattern: Final substituted_replace_pattern : INVITE sip:18452055545@\1
019029: Feb 20 15:01:11.733: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Passing substituted replace pattern
019030: Feb 20 15:01:11.733: //-1/xxxxxxxxxxxx/SIP/Info/sip_profiles_application_modify_req_uri: Req URI after modification : INVITE sip:18452055545@192.168.1.200:5060 SIP/2.0
019031: Feb 20 15:01:11.733: //6176/F644081C837B/SIP/Info/sipSPIUpdateCallEntry:
Call 6176 set InfoType to SPEECH
019032: Feb 20 15:01:11.737: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 56
019033: Feb 20 15:01:11.737: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 61
019034: Feb 20 15:01:11.737: //6176/F644081C837B/SIP/Info/sentInviteRequest: Sent Invite in state STATE_IDLE
019035: Feb 20 15:01:11.737: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteRequest: Transaction active. Facilities will be queued.
019036: Feb 20 15:01:11.737: //6176/F644081C837B/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
019037: Feb 20 15:01:11.737: //6176/F644081C837B/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
019038: Feb 20 15:01:11.737: //6176/F644081C837B/SIP/Info/sipSPICreateRtpSession: sess: 486A33A4 do_rtcp:0
019039: Feb 20 15:01:11.737: //6175/F644081C837B/SIP/Info/ccsip_update_peer_caps: Stream is NULL
019040: Feb 20 15:01:11.737: //6175/F644081C837B/SIP/Info/ccsip_update_peer_caps: gccb/stream is NULL, not updating now !!
019041: Feb 20 15:01:11.737: //6176/F644081C837B/SIP/Info/sipSPIUpdateGccb: Failure from the peer leg so no updations now!!
019042: Feb 20 15:01:11.741: //6176/F644081C837B/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:18452055545@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK18D411B7
From: <>18165297500@callcentric.com>;tag=1537CB50-19F9
To: <17772253754>
Date: Thu, 20 Feb 2014 21:01:11 GMT
Call-ID: F6503D05-99A811E3-8381E4C3-DC5CB4B2@192.168.1.203
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 4131653660-2577928675-2205934787-3697063090
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392930071
Contact: <18165297500>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309

v=0
o=CiscoSystemsSIP-GW-UserAgent 6875 1675 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18466 RTP/AVP 18 0 8 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

can you send me a webex invite..just post the link here..

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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whatuusay1
Level 1
Level 1

Wrap up... OK so its works now

I've attached the final running config - NOTE this doesnt have outbound working (that will require some more work)  but inbound with DID routing works

The attached logs

'final-debug.log' is the ccsip debug messages & info showing calling on 2 did's.

'debug voip ccapi inout' shows the dial-peer matching.

'final-running-config.log' is the running config with did matching - note that outbound with caller id isnt setup yet.

HUGE Thanks for Aok and everyone on the forum for the help.

- Andrew