SIPGATE to Cisco Callmanager 10.5 (With Cube - CRS1000v)

[martindickinson]

Hello All,

I have just setup a SIPGATE (SIP) account and have out bound calls working. I am using Callmanager 10.5 and CRS1000v (As cube), both built in VMWare 5.5.

My problem is with inbound calls from SIPGATE. I get invite 103 messages using 'debug ccsip messages' and nothing else.

I suspect this is with the inbound dial-peers and was wondering if anyone has set this up before?

I have attached a copy of the 'debug ccsip messages' output.

Cheers in advance for any help.

Martin.

[Vivek Batra]

Issue seems with called party number in Request-URI, you're getting actual called party number in To field instead. To avoid further troubleshooting, can you please check if it's possible to get called party number from SIPGATE in Request-URI....

- Vivek

[martindickinson]

Hi Vivek,

Many thanks for your assistance.

The called party number is '00441524881202' (Confirmed with SIPGATE yesterday).

From: "07775706183" <sip:07775706183@sipgate.co.uk>;tag=as020512ef

To: <sip:00441524881202@sipgate.co.uk>

Contact: <sip:07775706183@217.116.117.68:5060>

Cheers,

Martin.

[Vivek Batra]

Hi Martin,

Yup I know but SIPGATE is sending 2454146e0 in Request-URI which is not numeric and CUBE fetches called party number from Request-URI only. What I was asking you to check in SIPGATE to send 00441524481202 in Request-URI.....

- Vivek

[martindickinson]

Hi Vivek,

I have asked SIPGATE to make this change and am waiting for a reply. Was wondering if i can translate the invite on the cube if they cant change the invite?

Many thanks,

Martin.

[martindickinson]

Hello All,

SIPGATE are unable to change the sip-uri below:

Dec 9 18:32:53.256: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:2454146e0@192.168.0.160:5060 SIP/2.0
Record-Route: <sip:217.10.79.23;lr;ftag=as451c0acb>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as451c0acb>
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bK5a5f.24b5efa212acbf854eb9a6891e0b1669.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK5a5f.9f22958c324d04ebaa31b45b6650eadc.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK5a5f.4a7b57d3a561e51319f40890c87adcb1.0
Via: SIP/2.0/UDP 217.116.117.5:5060;branch=z9hG4bK01a78470
Max-Forwards: 67
From: "077989xxxxx" <sip:077989xxxxx@sipgate.co.uk>;tag=as451c0acb
To: <sip:00441524881202@sipgate.co.uk>
Contact: <sip:077989xxxxx@217.116.117.5:5060>
Call-ID: 448e4f7729039cdb1a7f8ccb5f38ae42@sipgate.co.uk
CSeq: 103 INVITE

Does anyone know how i can make the inbound call work please?

Cheers, Martin.

[Vivek Batra]

Hi Martin,

Yes there is an option to get called party number from To feild as well. Please Google SIP profile copy.

However CUBE first needs to match inbound dial peer before applying further call routing logic. We need to verify if cottect dial peer is being matched considering we don't have appropriate numeric called party number in Request-URI. Can you please share the output of debug voice ccapi so that we can verify the same.

- Vivek

[martindickinson]

Hi Vivek,

I get no output from debug voice ccapi.

Do you want me to attach config?

Martin.

[Vivek Batra]

Hi Martin,

Since you have configured incoming called-number 00441524881...$ in dial-peer voice 1 for inbound calls, this dial peer won't match since CUBE fetches called party number from Request-URI and SIPGATE is not sending actual number in Request-URI.

Can you change the command to incoming called-number . and see if debug voice ccapi inout gives you an output confirming that dial-peer 1 is being matched. Although I am in the opinion that still it will not work because of presence of character in Request-URI and widlcard in incoming called-number command is limited to matching numerals only.

If it works for you let us know so that we can apply further configuration (SIP profile) otherwise use URI based mechanism to match inbound dial peer. Please check the below article how to configure dial peers to match on the basis of URI's.

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-inbnd-dp-match-uri.html

I will suggest you to match on the basis of From field which will remain static viz sipgate.co.uk

Once we are able to match inbound dial peer, we will have to add some SIP profile commands to fetch the original called party number from To field for further call routing.

Please let me know if you have any question.

- Vivek

[martindickinson]

Hi Vivek,

I have made the change to the inbound dial-peer but no output from 'debug voice ccapi'

dial-peer voice 1 voip
 description ***Inbound WAN Side ***
 session protocol sipv2
 incoming called-number .
 voice-class codec 2
 dtmf-relay rtp-nte
 no vad

Do you know if the inbound 'Sip uri' can be transformed to be the 'Called number' instead of the Sipgate username?

Many thanks,

Martin.

[Vivek Batra]

Hi Martin,

Yes it's possible as I said before but only after matching inbound dial peer and since incoming called-number didn't help us in matching inbound dial peer, please configure URI based command to match dial peer. Once done, we can configure SIP profile to get the called number from TO filed.

Please let me know if you face any challenges in configuring the same.

- Vivek

[martindickinson]

Hi Vivek,

I have been through the document you sent and made some other changes. I now have the DDI number in my sip uri: (00441524881202).

Dec 11 20:11:03.936: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:00441524881202@192.168.0.160:5060 SIP/2.0
Record-Route: <sip:217.10.79.23;lr;ftag=as3bf399a2>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as3bf399a2>
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bKf163.7d73df562f542e1757ac9475319456bd.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bKf163.950864bc29b5790ea50be196f82b5055.1
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bKf163.9870b2b900f3853f4be2cda554b2ffbb.0
Via: SIP/2.0/UDP 212.9.44.6:5060;branch=z9hG4bK7e825502
Max-Forwards: 67
From: "07775xxxxxx" <sip:07775xxxxxx@sipgate.co.uk>;tag=as3bf399a2
To: <sip:00441524881202@sipgate.co.uk>
Contact: <sip:07775xxxxxx@212.9.44.6:5060>

I also have the following config on my inbound Dial-peer from sipgate.

dial-peer voice 1 voip
description ***Inbound WAN Side ***
translation-profile incoming INCOMING-SIP
session protocol sipv2
incoming called-number .
incoming uri via 100
voice-class codec 2
voice-class sip privacy-policy passthru
voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay rtp-nte
no vad
!

Debug voice ccapi - still shows no output.

Any ideas please?

Cheers Martin.

[martindickinson]

Hello Vivek,

I have attached my configs. The outbound works fine as discussed.

voice service voip
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
h225 display-ie ccm-compatible
sip
bind control source-interface GigabitEthernet1
bind media source-interface GigabitEthernet1
rel1xx disable
min-se 360 session-expires 360
header-passing
error-passthru
early-offer forced
!
voice class codec 1
codec preference 1 g729r8
!
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729br8
codec preference 4 g729r8
!
!
voice class sip-profiles 101
request INVITE sip-header From modify "<sip:(10..)@sipgate.co.uk>" "<sip:2454146e0@sipgate.co.uk>"
request REGISTER sip-header To modify "<sip:00441524881202@sipgate.co.uk>" "<sip:2454146e0@sipgate.co.uk>"
!
voice class sip-profiles 102
request INVITE sip-header SIP-Req-URI modify "<sip:2454146e0@192.168.0.160:5060 SIP/2.0>" "<sip:1021@192.168.0.160:5060 SIP/2.0>"
!
!
!
!
!
voice translation-rule 1
rule 1 /^00441524881202/ /1021/
!
!
voice translation-profile INCOMING-SIP
translate called 1
!
!
!
license udi pid CSR1000V sn 98FLUZ4LKD6
!
spanning-tree extend system-id
!
!
redundancy
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface GigabitEthernet1
ip address 192.168.0.160 255.255.255.0
negotiation auto
!
interface GigabitEthernet2
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet3
no ip address
shutdown
negotiation auto
!
!
virtual-service csr_mgmt
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
ip tftp source-interface GigabitEthernet1
ip dns server
ip route 0.0.0.0 0.0.0.0 192.168.0.1
!
!
!
!
control-plane
!
!
!
!
!
!
!
!
dial-peer voice 1 voip
description ***Inbound WAN Side ***
session protocol sipv2
incoming called-number 00441524881...$
voice-class codec 2
voice-class sip profiles 102
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10 voip
description ***Inbound 01524 881202***
shutdown
destination-pattern 10..
session protocol sipv2
session target ipv4:192.168.0.150
voice-class codec 2
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
dtmf-interworking rtp-nte
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 20 voip
description ***Outbound WAN side***
destination-pattern .T
session protocol sipv2
session target dns:sipgate.co.uk
incoming called-number .
voice-class codec 1
no voice-class sip privacy-policy passthru
voice-class sip options-ping 60
voice-class sip early-offer forced
voice-class sip profiles 101
!
dial-peer voice 5 voip
description ***Outbound LAN side to CUCM***
shutdown
destination-pattern 00441524881...$
session protocol sipv2
session target ipv4:192.168.0.150
voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay rtp-nte
codec g711ulaw
no vad

Many thanks,

Martin.