ISP SIP Trunk will not register

[ibbgermany]

Hello Community,

we switched from our PBX to SIP Trunk. We have a CISCO 29series (without CUBE). The router interface is directly connected to the Internet with a public IP address. Unfortunate we cannot register our SIP Trunk to the ISP. What did I wrong? Can you help me please? The Telephone number (example) is: +493012345

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

voice service voip
ip address trusted list
ipv4 x.x.x.x 255.255.0.0
ipv4 x.x.x.x.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
no supplementary-service h225-notify cid-update
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
h225 timeout call-proceeding 25
h225 signal overlap
call preserve
sip
session refresh
registrar server
asserted-id pai
outbound-proxy dns:reg.sip-trunk.telekom.de
conn-reuse
no update-callerid
early-offer forced
midcall-signaling passthru
privacy-policy passthru
sip-profiles inbound
!
!
voice class uri 101 sip
host ipv4:x.x.x.x
voice class codec 1
codec preference 1 g722-64
codec preference 5 g711alaw
!
!
!
voice class e164-pattern-map 201
e164 11[68]T
e164 11[025]
e164 +T
e164 0T
!
!
voice class server-group 1000
ipv4 x.x.x.x
description --- TO CUCM ---
!
voice class sip-options-keepalive 101
up-interval 30
retry 3
transport tcp
!
!
license udi pid CISCO2921/K9 sn FCZ184360K6
license accept end user agreement
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!

interface GigabitEthernet0/0
description --- router ip address ---
ip address x.x.x.x 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
description --- to internet public ip address---
ip address x.x.x.x
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
duplex auto
speed auto
!

dial-peer voice 201 voip
description **SIP-TRUNK.TELEKOM.DE**
session protocol sipv2
session target sip-server
session transport tcp
destination e164-pattern-map 201
incoming called-number .T
voice-class sip outbound-proxy dns:reg.sip-trunk.telekom.de
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
fax-relay ecm disable
fax rate 14400
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
ip qos dscp cs6 signaling
clid strip name
no vad
!
dial-peer voice 101 voip
description **CUCM**
destination-pattern +493012345T
session protocol sipv2
session target ipv4:X.x.x.x:5600
session transport tcp
incoming uri via 101
no voice-class sip outbound-proxy
voice-class sip options-keepalive profile 101
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
codec g711alaw
fax-relay ecm disable
fax rate 14400
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
!
sip-ua
credentials number +4930123450 username 55.......... password 7 ........ realm sip-trunk.telekom.de
authentication username 55.......... password 7 ........ realm sip-trunk.telekom.de
no remote-party-id
timers expires 900000
timers register 100
timers dns registrar-cache ttl
registrar dns:sip-trunk.telekom.de expires 240 tcp auth-realm sip-trunk.telekom.de
sip-server dns:sip-trunk.telekom.de
connection-reuse
!

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

[Rajan]

Hi,

What error you are seeing when you enable "debug ccsip messages" ?

[ibbgermany]

Hi Rajan,

sorry for the late response. We really appreciate your help.

Please see the attached ccsip messages.

[Rajan · Accepted solution]

It seems like the provider is rejecting the call with 403 Forbidden error message. You need to check with the provider on what they expect as ANI, DNIS and need to configure to send these details accordingly in the invite message (FROM, TO header) sent to them.

HTH
Rajan
Please rate all useful posts by clicking the star below and mark solutions as accepted wherever applicable

[ibbgermany]

Hi Rajan,

thank you so much. We'll check this with our ISP. In addition, we'll buy a new router ISR4321 including CUBE license.

[Chris Deren]

New router and CUBE licenses are not going to change the behavior, because your current 29XX should work fine, and CUBE licenses are not enforced.  You should still get a supported model as 29XX are EOS, and definitely have proper CUBE licenses, but if your thinking is that the replacement will resolve the issue don't expect that.  As suggested work with he provider on why they are rejecting the call.

[ibbgermany]

Hi Chris,

thank you so much for your info. Mmhh, I thought my old router is the major problem.I'll check this with our ISP. In addition I'll log the SIP via Wireshark.

Hopefully I can see here what's going on from my router to ISP.

I called them this week and they told me, that our main registered number is not registered/ they cannot see any number.... Strange.