12-17-2013 07:46 AM - edited 03-16-2019 08:54 PM
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
12-18-2013 03:09 AM
Can you send across your translation pattern and dial-peer config ?
Thanks
Manish
12-18-2013 04:27 AM
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
12-18-2013 05:12 AM
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.
12-18-2013 06:27 AM
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
12-18-2013 07:07 AM
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
12-18-2013 10:19 PM
I experience the same issue. I need a Role that looks into the "to" field.
So use this:
The general command that defines a rule to add a field to a SIP method/response is:
The general command that defines a rule to remove a field to a SIP method/response is:
The general command that defines a rule to modify a field to a SIP method/response is:
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!
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