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New Member

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

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

Everyone's tags (4)
1 ACCEPTED SOLUTION

Accepted Solutions
VIP Super Bronze

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

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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47 REPLIES

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

Hi,

Whatever i can see from CUBE config , you are sending called number 17772253754 to CUCM but CUBE did not invoked second call leg  i.e. from CUBE to CUCM.

Can you try by applying these changes...

dial-peer voice 6 voip

description ## INBOUND CALL from ITSP ##

session protocol sipv2

incoming called-number 17772253754

voice-class sip profiles 1

dtmf-relay h245-alphanumeric

no vad

dial-peer voice XXX voip

description ## INBOUND DID to CUCM ##

session protocol sipv2

session target ipv4:192.168.1.200

destination-pattern 17772253754

dtmf-relay h245-alphanumeric

no vad

And then send across "debug ccsip message" from the cube.

Thanks

Manish

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

Hi.

In addition to what correctly suggested by Manish (+5), please modify dtmf mode on sip dialpeers to rtp-nte.

HTH

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

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

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

Thank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.

dial-peer voice 6 voip

description ## INBOUND CALL from ITSP ##

session protocol sipv2

session target sip-server

incoming called-number 17772253754

voice-class sip profiles 1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 100 voip

description ## INBOUND DID to CUCM ##

destination-pattern 17772253754

session protocol sipv2

session target ipv4:192.168.1.200

voice-class sip profiles 1

dtmf-relay rtp-nte

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 rtp-nte

no vad

Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: <18165297500>;tag=3601469891-655
t: <>18452055544@ss.callcentric.com>
i: 3184639-3601469891-623@msw2.telengy.net
CSeq: 1 INVITE
Max-Forwards: 8
m: <0778BAA027205CDCA5AED80C59F25866>
Supported: timer
c: application/sdp
l: 350

v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
s=sip call
c=IN IP4 204.11.192.164
t=0 0
m=audio 61782 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 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: <18165297500>;tag=3601469891-655
To: <>18452055544@ss.callcentric.com>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: 3184639-3601469891-623@msw2.telengy.net
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Feb 15 10:18:11.460: //2420/9933162D820E/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=z9hG4bK9A91F35
From: <>18165297500@callcentric.com>;tag=8408644-12C8
To: <17772253754>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: 993622F5-959311E3-8214FFD7-951695DD@callcentric.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392481091
Contact: <18165297500>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
f: <>18165297500@callcentric.com>;tag=8408644-12C8
t: <17772253754>
i: 993622F5-959311E3-8214FFD7-951695DD@callcentric.com
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
l: 0


Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:17772253754@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: <>18165297500@callcentric.com>;tag=8408644-12C8
To: <17772253754>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: 993622F5-959311E3-8214FFD7-951695DD@callcentric.com
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Feb 15 10:18:11.560: //2420/9933162D820E/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=z9hG4bK9AA1BA3
From: <>18165297500@callcentric.com>;tag=8408644-12C8
To: <17772253754>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: 993622F5-959311E3-8214FFD7-951695DD@callcentric.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392481091
Contact: <18165297500>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:17772253754@192.168.1.200:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273

v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
f: <>18165297500@callcentric.com>;tag=8408644-12C8
t: <17772253754>
i: 993622F5-959311E3-8214FFD7-951695DD@callcentric.com
CSeq: 102 INVITE
l: 0


Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:17772253754@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: <>18165297500@callcentric.com>;tag=8408644-12C8
To: <17772253754>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: 993622F5-959311E3-8214FFD7-951695DD@callcentric.com
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0


Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: <18165297500>;tag=3601469891-655
To: <>18452055544@ss.callcentric.com>;tag=8408714-B60
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: 3184639-3601469891-623@msw2.telengy.net
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0


Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

apsc-vrtr1#ACK sip:17772253754@192.168.1.203:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: <18165297500>;tag=3601469891-655
t: <>18452055544@ss.callcentric.com>;tag=8408714-B60
i: 3184639-3601469891-623@msw2.telengy.net
CSeq: 1 ACK
Max-Forwards: 10
l: 0


vrtr1#u al
Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;npdi@alpha16.callcentric.com:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
f: <18165297500>;tag=3601469891-655
t: <>18452055544@ss.callcentric.com>;tag=8408714-B60
i: 3184639-3601469891-623@msw2.telengy.net
CSeq: 1 ACK
Max-Forwards: 8
l: 0

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

HI

Can you check your authentication under sip-ua?.Check the username and password under sip-ua.

SIP/2.0 403 Incorrect Authentication.

Thanks

please rate all useuful information

VIP Super Bronze

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

THe new INVITE you are sending doesnt contain any number in the RURI..

This is the INVITE sent to CUCM

Sent:

INVITE sip:@192.168.1.200:5060 SIP/2.0 (there is no user portion in the INVITE..i.e the called number)

Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3

From: <>18165297500@callcentric.com>;tag=8408644-12C8

I suggest you remove the sip profile 1 from your dial-peer to cucm because I think its messing up your INVITE

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New Member

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

Thank you - I removed the sip profile from the dial-peers as a troubleshooting step. I've attached the ccsip messages debug after doing so.

It looks like the SIP INVITE now has a number (its the username and not the actual did i called). It looks like callcentric replied with a SIP/2.0 407 Proxy Authentication Required. Not sure how to get past that..

INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

Config as configured with the ccsip message debug.

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

dial-peer voice 6 voip

description ## INBOUND CALL from ITSP ##

session protocol sipv2

session target sip-server

incoming called-number 17772253754

voice-class codec 1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 100 voip

description ## INBOUND DID to CUCM ##

destination-pattern 17772253754

session protocol sipv2

session target ipv4:192.168.1.200

voice-class codec 1

dtmf-relay rtp-nte

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 codec 1

dtmf-relay rtp-nte

no vad

sip-ua

credentials username 17772253754 password 7 047919561B29495C48 realm callcentric.com

authentication username 17772253754 password 7 106C1B49111F17195C5C realm 24.123.98.94

no remote-party-id

retry invite 4

retry response 3

retry bye 2

retry register 5

timers connect 100

mwi-server dns:callcentric.com expires 1800 port 5060 transport udp

registrar dns:callcentric.com expires 1800

sip-server dns:callcentric.com:5060

host-registrar

Feb 15 22:15:28.440: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-c0773d290cf0723fce82a45842ed1a9d
f: <18165297500>;tag=3601512928-48129
t: <>18452055544@ss.callcentric.com>
i: 3299487-3601512928-48097@msw2.telengy.net
CSeq: 1 INVITE
Max-Forwards: 8
m: <33E88838C1C1F83FE4D98EEEC68D9BA0>
Supported: timer
c: application/sdp
l: 350

v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
s=sip call
c=IN IP4 204.11.192.164
t=0 0
m=audio 63994 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 15 22:15:28.456: //3192/CD429E0782C5/SIP/Msg/ccsipDisplayMsg:
Sent:

apsc-vrtr1#SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-c0773d290cf0723fce82a45842ed1a9d
From: <18165297500>;tag=3601512928-48129
To: <>18452055544@ss.callcentric.com>
Date: Sun, 16 Feb 2014 04:15:28 GMT
Call-ID: 3299487-3601512928-48097@msw2.telengy.net
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Feb 15 22:15:28.592: //3193/CD429E0782C5/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKCC913E1
From: <>18165297500@callcentric.com>;tag=AD13394-1090
To: <17772253754>
Date: Sun, 16 Feb 2014 04:15:28 GMT
Call-ID: CD46E3B8-95F711E3-82CBFFD7-951695DD@callcentric.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3443695111-2515997155-2194014167-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392524128
Contact: <18165297500>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 284

v=0
o=CiscoSystemsSIP-GW-UserAgent 2039 178 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 17858 RTP/AVP 18 0 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Feb 15 22:15:28.688: //3193/CD429E0782C5/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKCC913E1;rport=64232;received=24.123.98.94
f: <>18165297500@callcentric.com>;tag=AD13394-1090
t: <17772253754>
i: CD46E3B8-95F711E3-82CBFFD7-951695DD@callcentric.com
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="a3004fd6e15e380e2387317080d640fe", opaque="", stale=TRUE, algorithm=MD5
l: 0


Feb 15 22:15:28.700: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:17772253754@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKCC913E1
From: <>18165297500@callcentric.com>;tag=AD13394-1090
To: <17772253754>
Date: Sun, 16 Feb 2014 04:15:28 GMT
Call-ID: CD46E3B8-95F711E3-82CBFFD7-951695DD@callcentric.com
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Feb 15 22:15:28.700: //3192/CD429E0782C5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-c0773d290cf0723fce82a45842ed1a9d
From: <18165297500>;tag=3601512928-48129
To: <>18452055544@ss.callcentric.com>;tag=AD13404-1D88
Date: Sun, 16 Feb 2014 04:15:28 GMT
Call-ID: 3299487-3601512928-48097@msw2.telengy.net
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=47
Content-Length: 0


Feb 15 22:15:28.784: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

vrtr1#ACK sip:17772253754@192.168.1.203:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-c0773d290cf0723fce82a45842ed1a9d
f: <18165297500>;tag=3601512928-48129
t: <>18452055544@ss.callcentric.com>;tag=AD13404-1D88
i: 3299487-3601512928-48097@msw2.telengy.net
CSeq: 1 ACK
Max-Forwards: 10
l: 0


vrtr1#u all
All possible debugging has been turned off

VIP Super Bronze

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

Before we deal with the issue of the called number, lets deal with this signle call first. CUBE is sending service unavailable..Ledts deal with that first..Configure the following and attach the debug

conf t

voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw

codec preference 3 g711alaw---------------------add this

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then..

debug ccsip all


(such as Putty)

then do the ff:

terminal length 0
show logging

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New Member

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

Thank you - I've attached the requested ccsip all debugs.

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

If i am correct 192.168.1.200 is your CUCM ip.

Here CUCM is responding with proxy authentication required message. I dont know why CUCM is asking for proxy authentication, did you added SIP trunk on your CUCM correctly ?  Does any kind of authentication placed in CUCM sip trunk for Inbound calls ?

Sent:

INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKCC913E1

From: <>18165297500@callcentric.com>;tag=AD13394-1090

To: <17772253754>

Date: Sun, 16 Feb 2014 04:15:28 GMT

Call-ID: CD46E3B8-95F711E3-82CBFFD7-951695DD@callcentric.com

Received:

SIP/2.0 407 Proxy Authentication Required

v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKCC913E1;rport=64232;received=24.123.98.94

f: <>18165297500@callcentric.com>;tag=AD13394-1090

t: <17772253754>

i: CD46E3B8-95F711E3-82CBFFD7-951695DD@callcentric.com

Now CUBE is responding to that proxy authentication request with Callcentric credentials which is incorrect. The credentials for callcentric is only for the call placed or received from ITSP not towards the CUCM.

Cal you attach "debug voice dialpeer inout" for a test call.

Rate all the helpful posts.

Thanks

Manish

VIP Super Bronze

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

Manish,

CUBE is not responding to the proxy authentication request at all...It just sent Service unavailable with cause code 47

Please rate all useful posts

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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

Oh.Yes ..Correct....I was just referring to the second "debug ccsip message" after adding the dial-peer in which CUBE respond with callcentric credentials.

I think that has been fixed by removing sip profile from dial-peer.

Thanks

Manish

New Member

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

I've attached the dial-peer debugs as requested - more information if nothing else.

Thank you,

Andrew

VIP Super Bronze

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

Okay, the logs show that CUBE cant handle the authentication challenge cucm is sending..

Can you go to cucm and disable digest authentication on your sip security profile for the trunk...lets start from there

004483: Feb 15 23:38:10.368: //3285/5A9EFEB982E8/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials

004484: Feb 15 23:38:10.368: //3285/5A9EFEB982E8/SIP/Error/act_sentinvite_new_message: Error handling AuthenticationChallenge

004485: Feb 15 23:38:10.368: //3285/5A9EFEB982E8/SIP/State/sipSPIChangeState: 0x4963B838 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)

004486: Feb 15 23:38:10.368: //3285/5A9EFEB982E8/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:47, category:181

004487: Feb 15 23:38:10.368: //3285/5A9EFEB982E8/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(47) for outgoing call

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

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New Member

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

I checked the default Non Secure SIP Trunk Profile and

Thank you,

Andrew

New Member

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

Here are the sip profile screenshots.

Thank you,

Andrew

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

Can you reset SIP trunk once and then test a call.

Thanks

Manish

New Member

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

I reset the trunk and retested (same failure) - I've attached another set of ccsip all debugs (post sip trunk reset).

Thank you,

Andrew

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

Received:

SIP/2.0 407 Proxy Authentication Required

v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKD58157A;rport=49552;received=24.123.98.94

--------------------------------------------------------------------------------------------------------

Can you also remove session target sip-server from dial-peer 6.

dial-peer voice 6 voip

description ## INBOUND CALL from ITSP ##

session protocol sipv2

session target sip-server

Thanks

Manish

VIP Super Bronze

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

I have taken some time to look in detail into the logs and here is my summary...

When you configure ooutbound-proxy dns:callcentric.com, all outbound SIP request is sent to that domain (call centric.com)

you need to remove the command "outbound-proxy dns:callcentric.com"

the summary below  explain why

++receive INVITE from ITSP+++

Received:

INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-11f85db5478eab1a1d6560665917bdc9

f: <18165297500>;tag=3601520592-546043

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

i: 3308051-3601520592-546012@msw2.telengy.net

+++Next CUBE sends a trying to ITSP..Before it does that it needs to establish a TCP connection+++

NB: CUBE has sent the call back to ITSP on port 5080, because thats the port used for the incoming connection+++

008382: Feb 16 00:23:12.949: //3342/A5A3315A8319/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x49E83E84 to default port=5080

008383: Feb 16 00:23:12.949: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:204.11.192.159, rport:5080 with laddr:192.168.1.203

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-11f85db5478eab1a1d6560665917bdc9

From: <18165297500>;tag=3601520592-546043

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

+++Next CUBE needs to send an INVITE out to CUCM+++

NB: The target host and the outbound Host are different...The target host is who the call is for, the outbound host is who the call should be sent through+++

08432: Feb 16 00:23:12.965: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : 192.168.1.200 target_port : 5060

008433: Feb 16 00:23:12.965: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: outbound_host : callcentric.com outbound_port : 5060

++++Because your outbound_host is a dns entry, cube needs to resolve the DNS for the domain+++

008547: Feb 16 00:23:12.981: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE

008548: Feb 16 00:23:12.981: //3343/A5A3315A8319/SIP/State/sipSPIChangeState: 0x49630448 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_SENT_DNS)

+++CUBE does a SRV DNS query (looks like your DNS is configured for SRV++++++++++++

008554: Feb 16 00:23:12.985: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.callcentric.com and type:1

++++Next DNS server returns all the SRV records for _sip._udp.callcentric.com++++++++

008555: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha18.callcentric.com

008556: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008557: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha19.callcentric.com

008558: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008559: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha11.callcentric.com

008560: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008561: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha12.callcentric.com

008562: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008563: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha15.callcentric.com

008564: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008565: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha16.callcentric.com

008566: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008567: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha17.callcentric.com

008568: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008569: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Selected Server is alpha18.callcentric.com

008570: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: TYPE A query successful for alpha18.callcentric.com

008571: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: ttl for A records = 263 seconds

+++Next DNS sends the ip address for the chosen device..204.11.192.170 (alpha18.callcentric.com)++

008572: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: IP Address of alpha18.callcentric.com is:

008573: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: 204.11.192.170

+++NEXT CUBE sends a UDP connection to 204.11.192.170++++++++++

008620: Feb 16 00:23:13.077: //3343/A5A3315A8319/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer

008622: Feb 16 00:23:13.077: //3343/A5A3315A8319/SIP/Transport/sipSPITransportSendMessage: msg=0x4A9AEC54, addr=204.11.192.170, port=5080, sentBy_port=0, local_addr=192.168.1.203

008626: Feb 16 00:23:13.077: //-1/xxxxxxxxxxxx/SIP/Transport/sipCreateConnHolder: Created new holder=0x4A9B0760, addr=204.11.192.170; nailed=FALSE

008627: Feb 16 00:23:13.077: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostRequestConnection: Posting UDP conn create request for addr=204.11.192.170, port=5080, context=0x4840BD14

+++CUBE succesfully created a connection with 204.11.192.170+++++++++++

008638: Feb 16 00:23:13.081: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessConnCreated: connection instance created for addr:204.11.192.170, port:5080 local_addr=192.168.1.203 local_port=49552

Next CUBE procceeds to send an outbound INVITE

008646: Feb 16 00:23:13.081: //3343/A5A3315A8319/SIP/State/sipSPIChangeState: 0x49630448 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_SENT_INVITE, SUBSTATE_NONE)

++++++++++++but the INVITE appears to go to cucm when infact it wasnt sent there..it was sent to ITSP!!!!!!!!!!

Sent:

INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKD58157A

From: <>18165297500@callcentric.com>;tag=B46262C-230A

To: <17772253754>

++++NEXT we get a response back from your ITSP requesting authentication++++++++

008670: Feb 16 00:23:13.181: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog

008671: Feb 16 00:23:13.181: //3343/A5A3315A8319/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKD58157A;rport=49552;received=24.123.98.94

f: <>18165297500@callcentric.com>;tag=B46262C-230A

t: <17772253754>

i: A5AF6643-960911E3-831FFFD7-951695DD@callcentric.com

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="b69582701157bab55830417b7b62def9", opaque="", stale=TRUE, algorithm=MD5

l: 0

So, even though it looks like the call was sent to cucm, it was never sent to CUCM, it was sent to ITSP for onward forwarding to CUCM!!!!!!!!!!!!!!!!

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

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

Aok - Hugely helpful notes!! you were right - I removed the "outbound-proxy dns:callcentric.com" command and calls now ring through to my CUCM registered SCCP phone. When I answer the CUCM phone I receive a couple seconds of silence and then reorder tones. The calling party (18165297500) receives ringback and then an error that the party isnt avaialble from the local telco.

I've attached ccsip all debugs and cucm traces for the failed call as well as a current running config from the cube gateway.

Thank you so much for the assistance thus far - we're making progress

Calling number is 18165297500

Called number is 8452055544 (although the callcentric username 17772253754 is what you'll see in the logs)

Time of call 1:24PM CST

SCCP Phone Mac 001EBE91435C

Thanks,

Andrew

VIP Super Bronze

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

Excellent..We are making proigress...lookslike you dont like me though You havent rated any of my posts! My last post took all of my brains to figure out ! lol

Okay, Now the call progressed as expected and CUBE sends a 200 OK back to your ITSP..but they didnt respond to the 200 OK. You need to get on the phone with them and find out why they arent sending an ACK to your 200 OK. Come back and update us once thats done. We will then work on the sip profile to sort out the how to copy the TO header to the Request URI

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

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

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

Aok - My appologies, I think i didnt understand how the rating was working (I was rating the wrong posts). I rated your responses - very very helpful :-) Thank you!

I sent a note to the ITSP (Callcentric) and will respond once I hear back from them. They do not have a CUCM/CUBE template but they have been helpful in troubleshooting. They too saw the reinvite back (from the proxy statement) but weren't sure how to explain it.

It makes me happy it was a tough problem - I hate to ask for help on th easy ones

Thanks again and will reply shortly.

Andrew

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

Deji

My last post took all of my brains to figure out -  +5 for this.

New Member

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

AOK - I received the following back from the ITSP. It looks like they dont like the require timer statement.

From the trace provide we do see the following:

Require: timer

Can you please modify your configuration so that you are no longer requiring session timers? Once you have done so retest calling then update this ticket.

If you have any other questions, please feel free to ask. Thank you.

Thank you,

Andrew

VIP Super Bronze

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

Ok..Try this...and test again. (if it doesnt work please send debug ccsip messages only)

voice class sip-profiles 5

response 200 sip-header Require REMOVE

voice service voip

sip

sip-profiles 5

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New Member

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

Aok - We have sucess! Inbound calls are now sucessful! Talk about a happy day..I still have the DID routing issue, and i need to get outbound calls to include username or a DID on my account to sucessfully route .. but I dont think ive ever been so happy to hear a test call go through.

Here's the info they provide for outbound Caller ID.

To send the caller ID number of a DID on your account or an already verified number within the SIP INVITE message of an outbound call you will need to have your user agent (UA) attach any of the following headers from highest priority to lowest priority, which are supported by most IP PBX's (check your vendor's documentation for support), to your outbound calls:

REMOTE-PARTY-ID
P-ASSERTED-IDENTITY
P-PREFERRED-IDENTITY

Thank you,

Andrew

VIP Super Bronze

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

For the inbound call..try this...and test to see that calls are sent using the TO header..

voice class sip-profiles 5

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

For outbound calls, I need to know the range of your DDI...What should the extensions map to or do you just want to present a single DDI to your provider for all calls? If that is correct, then what is the DDI you want to use

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

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New Member

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

  Aok - Thanks, I'll give that a shot and see if the calls route correctly (I wont be able to test until this evening).

I have the following DID's

18452055544

18452055545

Outbound I'd want to see where the call was coming from and correctly insert that into the header - I assume something like

sip profile 6

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

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

And then put that on the outbound dial-peer to Callcentric (ITSP)?

Thanks again for all the assistance

VIP Super Bronze

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

voice class sip-profile 6

request INVITE sip-header P-Asserted-Identity modify “(.*)@(.*)” "18452055544@\2"

request INVITE sip-header Remote-Party-ID modify “(.*)@(.*)” "18452055544@\2"

Then apply to dial-peer towards ITSP or apply it to existing sip profile 5

sip profiles 5

request INVITE sip-header P-Asserted-Identity modify “(.*)@(.*)” "18452055544@\2"

request INVITE sip-header Remote-Party-ID modify “(.*)@(.*)” "18452055544@\2"

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