CUBE TO: Address not used in outgoing Dialpeer to CUCM

mjuch · ‎12-17-2013

Hello, we use a CUBE between Provider and CUCM.

Within incomming Invites we received the "Blocknumer without DDI" in the Header and the Number with DDI in the TO: Address.

The outgoing Dial-peer matches and send the Call to CUCM but only with the Blocknumber without DDI. It seems that the TO: Address is not used.

Is this configurable??

br Mike

incomming Invite

Received:
INVITE sip:08912345@212.202.129.51:5060 SIP/2.0
Via: SIP/2.0/UDP 213.148.136.218:5060;branch=z9hG4bK20cbo9309811o85t37l1.1
Call-ID: SDfa23e01-1497777e363b30dd0ca9b75e31675e26-l65h8l3
From: <sip:004921112345@qsc.de;user=phone>;tag=SDfa23e01-70ahhhs8-CC-34
To: <sip:0891234515@qsc.de;user=phone>
CSeq: 1 INVITE
Max-Forwards: 63
Contact: <sip:004921187554138@213.148.136.218:5060;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
Content-Length: 321
Content-Type: application/sdp
P-Called-Party-ID: <sip:0891234515@qsc.de>
X-ORIGINAL-DDI-URI: sip:0891234515@qsc.de
X-ORIGINAL-DDI-USER: 0891234515
X-CID: l0a0s75eae8hs05qnsarlhq18lla10n0@SoftX3000

v=0
o=HuaweiSoftX3000 66041570 66041570 IN IP4 213.148.136.218
s=Sip Call
c=IN IP4 213.148.136.218
t=0 0
m=audio 37882 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
a=fmtp:18 annexb=yes

Invite to CUCM

Sent:

INVITE sip:08912345@10.32.8.21:5060 SIP/2.0

Via: SIP/2.0/UDP 212.202.129.51:5060;branch=z9hG4bK281C98

From: <sip:004921187554138@213.148.136.218>;tag=D52E94-C32

To: <sip:08912345@10.32.8.21>

Date: Tue, 17 Dec 2013 14:15:46 GMT

Call-ID:

8E4752FC-665C11E3-8028EC8D-417B9213@10.32.8.1

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

Min-SE: 90

Cisco-Guid: 2386959020-1717309923-2149772429-1098617363

User-Agent: Cisco-SIPGateway/IOS-15.3.2.T1

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

CSeq: 101 INVITE

Timestamp: 1387289746

Contact: <sip:004921187554138@212.202.129.51:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 62

Manish Prasad · ‎12-18-2013

Can you send across your translation pattern and dial-peer config ?

Thanks

Manish

mjuch · ‎12-18-2013

Hello Manish, i used a voip Dialpeer with destination-Pattern match

dial-peer voice 10 voip

destination-pattern 89T

session target ipv4:

session Protocoll sipv2

session transport udp

codec g711a

as we use a voip dialpeer all digits should be send.

In the debug voip dialpeer default, i can see that within the alalysis the cube noticed the number with DDI but at the end of the Analysis the Number without DDI is used as called Number. May it is a Bug. I expected that the cube/dialpeer use the TO Address.

may you have additional ideas?

br Mike

Manish Prasad · ‎12-18-2013

Are you sure dial-peer 10 is matching for inbound call towards cucm ?

Bcoz the number you are getting from ITSP is with zero - 0891xxxx but the dial-peer destination pattern is 89T.

Can you check any other explicit dial-peer matching for this pattern ? or send us debug voice dialpeer all.

mjuch · ‎12-18-2013

Hello Manish, it was a copy n'paste error. The dest pattern is 089T. The call is routed to cucm. This works, only the calling party number is not correct. In the moment i have only 2 dialpeers, one for incomming and one for outgoing. Best regards Mike

Manish Prasad · ‎12-18-2013

You certainly do modify sip messages through sip profile , check this link. -

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080982499.shtml#sip_messages - but that need some good expertise (which i don't have) . In this case i would recommend have a word with your telco that you are not receive whole DID number as dialed and send them sip logs.

Rate all the helpful post.

Thanks

Manish

armin.wagenfuehr · ‎12-18-2013

I experience the same issue. I need a Role that looks into the "to" field.

So use this:

General Configuration Command Structure

The general command that defines a rule to add a field to a SIP method/response is:

add

The general command that defines a rule to remove a field to a SIP method/response is:

remove

The general command that defines a rule to modify a field to a SIP method/response is:

modify

Configuration Steps

The first step is to define the rules. In order to define the rules, use the general command structure given in the previous section. For example:

voice class sip-profiles 100
  request INVITE sip-header…
  response 100 sip-header…
  request INVITE sdp-header…

The second step is to apply the rules either to the global or dial-peer level of the CUBE configuration. In order to apply the rules globally to all calls traversing CUBE, use this command structure:

voice service voip 
   sip
     sip-profiles 100

In order to apply the rules selectively to calls traversing only a particular outgoing dial-peer, use this command structure:

dial-peer voice 555 voip
   voice-class sip-profiles 100

HTH, please rate useful posts!

HTH, please rate all useful posts and right answers.