topic Re: SIP Normalization Script to replace Anonymous ID to ==== callerid in Call Control

SIP Normalization Script to replace Anonymous ID to ==== callerid

monasir — Tue, 27 Nov 2018 09:57:07 GMT

Hi ,

I need help with creating a SIP Normalization Script to replace Anonymous ID to caller ID's

The thing is, when i make a call (from external) to cucm and the extension has forwarderd all calls to another externally number for example ( mobile) the call is bein rejected.

When i do a debug ccsip message i'm seeing something like this in the output:

( From: "Anonymous" <sip:anonymous@voip.XXX.XXX>;tag=XXX.XX.XX.XX+1+4d22ea09+c7ab9eef )

So i simply want to replace the From field to a caller id so the calls will get true.

Normal calls without forwarded to other extensions are working fine and are not rejected, howe ever i still see in the output

From: "Anonymous" <sip:anonymous@voip.XXX.XXX>;tag=XXX.XX.XX.XX+1+4d22ea09+c7ab9eef )

I appreciate all the assistance!

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

Raghavendra G V — Thu, 29 Nov 2018 08:33:15 GMT

Please check below link might help you.
https://community.cisco.com/t5/ip-telephony-and-phones/normalization-script/m-p/2131611

Thanks,
Raghavendra

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

monasir — Thu, 29 Nov 2018 08:35:50 GMT

Hi,

I have seen this post, unfortunately this will not work for me.

I'm using cube

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

Raghavendra G V — Thu, 29 Nov 2018 09:07:02 GMT

are you looking for TCL IVR script on cube? or SIP Normalization script?

Thanks,

Raghavendra

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

monasir — Thu, 29 Nov 2018 09:14:05 GMT

Hi,

I'm looking for a SIP normalization script.

The thing is when calling from external , calls arriving in the cube with:

From: "Anonymous" <sip:anonymous@xx.xx.net>;

This is something that we do not want.

We want that calling number needs to be sent with calling id.

We have many issues like SNR, also we are not able to forward calls to other external numbers.

This is all because of the fact the calls arrive in the cube with anonymous, the provider is rejecting this .

Because it probably sees its not something valid , not a valid number withing the DID range.

Nowe it seems to be possible to make a SIP normalization script to change the from field with a valid number.

I'm a little bit lost because ive tried several workarounds with no succes.

any help would be appreciated

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

schaef350 — Tue, 11 Dec 2018 16:27:51 GMT

I would do this in CUBE before it ever gets to CUCM. Try something like this:

http://technologyordie.com/calling-party-routing-of-anonymous-calls-sip-header-fix-up

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

monasir — Wed, 12 Dec 2018 09:11:32 GMT

Hi i've tried the workaround, unfortunately it didnt work for me.

While debugging i still see Anonymous

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

schaef350 — Wed, 12 Dec 2018 15:51:27 GMT

Would you mind sharing your config?

Re: SIP Normalization Script to replace Anonymous ID to ==== callerid

monasir — Thu, 13 Dec 2018 12:31:28 GMT

Hi Schaef,

pfa running config

===========================================================================

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2018.12.10 10:49:24 =~=~=~=~=~=~=~=~=~=~=~=

CUBE-SCN00214#
CUBE-SCN00214#
CUBE-SCN00214#
CUBE-SCN00214#
CUBE-SCN00214#term len
CUBE-SCN00214#term length 0
CUBE-SCN00214#sh run
CUBE-SCN00214#sh running-config
Building configuration...

Current configuration : 7377 bytes
!
! Last configuration change at 05:38:41 AST Mon Dec 10 2018 by voippoc
! NVRAM config last updated at 07:00:59 AST Wed Dec 5 2018 by voippoc
!
version 15.5
service timestamps debug datetime localtime
service timestamps log datetime localtime
service password-encryption
no platform punt-keepalive disable-kernel-core
!
hostname CUBE-SCN00214
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
logging buffered 1000000
!
no aaa new-model
clock timezone AST -4 0
!
!
!
!
!
!
!
!
!
!
!

ip host voip.test.net 123.45.67.89
ip name-server xx.xx.xx.xx xx.xx.xx.xx

ip domain name xx.nl
!
!
!
!
!
!
!
!
!
!
subscriber templating
vtp mode transparent
multilink bundle-name authenticated
!
!
!
!
!
!
!
!
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
ip address trusted list
ipv4 12.34.5610 255.255.255.255
ipv4 12.34.5620 255.255.255.255
ipv4 123.45.67.89 255.255.255.255
no ip address trusted authenticate
address-hiding
mode border-element license capacity 200
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol pass-through g711alaw
sip
header-passing
registrar server expires max 3600 min 120
early-offer forced
midcall-signaling passthru
!
!
voice class uri InboundCUCM sip
host ipv4:12.34.5610
host ipv4:12.34.5620
!
voice class uri InboundSIPSP sip
host ipv4:123.45.67.89
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
!
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g729br8
!
!
!
voice class server-group 1
ipv4 12.34.5610 preference 1
ipv4 12.34.5620 preference 2
!
voice class server-group 2
ipv4 123.45.67.89 preference 1
!
voice class sip-options-keepalive 1
description TCP Options CUCM
down-interval 10
retry 3
transport tcp
!
voice class sip-options-keepalive 2
description UDP Options SIP SP
down-interval 10
retry 3
transport udp
!
voice class dial-peer provision-policy 10
preference 1 called calling
preference 2 to
!
voice class dial-peer provision-policy 20
preference 1 called calling
preference 2 to
!
!
!
!
!
!
!
voice-card 0/4
no watchdog
!
license udi pid ISR4321/K9 sn FDO22360K8B
!
spanning-tree extend system-id
!
username voippoc privilege 15 password 7 044D040F1F31434D5D4951
!
redundancy
mode none
!
!
vlan internal allocation policy ascending
!
!
!
!
!
!
interface GigabitEthernet0/0/0
description naar internet
ip address xx.xx.xx.xx 255.255.255.248
negotiation auto
!
interface GigabitEthernet0/0/1
description intern
ip address xx.xx.xx.xx 255.255.255.0
negotiation auto
!
interface Service-Engine0/4/0
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
negotiation auto
!
interface Vlan1
no ip address
shutdown
!
ip default-gateway xx.xx.xx.xx
ip forward-protocol nd
no ip http server
no ip http secure-server

!
!
!
!
!
control-plane
!
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0/1
sccp ccm 12.34.5610 identifier 1 priority 2 version 7.0
sccp ccm 12.34.5620 identifier 2 priority 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0/1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register test-MTP2
associate profile 1 register xx-MTP1
!
!
!
dspfarm profile 1 mtp
codec g711ulaw
codec pass-through
maximum sessions software 100
associate application CUBE
!
dspfarm profile 2 mtp
codec g711ulaw
codec pass-through
maximum sessions software 100
associate application SCCP
!
dial-peer voice 10 voip
description Inbound from CUCM
session protocol sipv2
session transport tcp
incoming uri via InboundCUCM
voice-class codec 1
dtmf-relay rtp-nte sip-notify sip-kpml
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 11 voip
description Outbound to SIPSP
destination-pattern T
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 2
voice-class sip options-keepalive profile 2
dtmf-relay rtp-nte sip-notify sip-kpml
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 20 voip
description Inbound from SIPSP
session protocol sipv2
session transport tcp
incoming uri from InboundSIPSP
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte sip-notify sip-kpml
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 21 voip
description Outbound to CUCM
destination-pattern 123456..
session protocol sipv2
session transport udp
session server-group 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte sip-notify sip-kpml
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
!
sip-ua
xx
xx
xx
net
retry invite 2
retry register 10
timers connect 100
timers register 100
registrar dns:voip.test.net expires 300 auth-realm voip.test.net
sip-server dns:voip.test.net
host-registrar
presence enable
!
!
line con 0
login local
stopbits 1
line aux 0
stopbits 1
line vty 0 4
privilege level 15
login local
transport input ssh
line vty 5 15
privilege level 15
login
transport input ssh
!
ntp server 192.168.100.97
ntp server 192.168.100.98
!
end