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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

Andrew,

Good stuff..Glad we could help. You  might want to mark the thread as answered..It help others to know what the resolution is

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

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Hi all.
Deji, great great and again great job!!!!
Just for this case you should receive the VIP designation for at least next 10 years my friend :)

Andrew, you have been so lucky to have been follwed by that man, because you can be sure that your issue is solved if he puts his hands on.

Again Congrats!!!

Now, just out of curiosity, which was the solution? :)

Cheers

Carlo

Sent from Cisco Technical Support iPhone App

Please rate all helpful posts "The more you help the more you learn"

Carlo, thank you my friend for the kind words..

There were three major issues...

1. sip outbound-proxy issue where all calls orginating from the CUBE was sent back to the provider even though the sip headers looked like they were sent to CUCM..You can read the analysis of that on the thread with this time stamp

16-Feb-2014 15:02                             (in response to whatuusay1) 

2. The ITSP didnt specify session timer, hence CUBE was sending a Require header in the 200 OK and they didnt like it. So I removed the Require header in the 200 Ok response:

response 200 sip-header Require remove

3. The third issue was routing calls based on the To header. The ITSP was sending the called number in the To header and the authentication DDI in the RURI..This was rather complicated due to the voice translation rules messing up the sip profiles.. You can see the solution on this time stamp..

20-Feb-2014 21:06                             (in response to Ayodeji oladipo Okanlawon)

The following sip profiles were used..

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 class sip-copylist 1

sip-header TO

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

Please rate all useful posts