CUBE CallRouting Issues

[bosechristoph]

Dear all,

I am currently setting up a CUBE (ISR 4321) in order to connect to the ISP's SIP Trunk. I can establish connections to the outside world, but seemingly can't get it to work routing calls to the internal phones.

I'd be happy about any advice regarding the dial-peers.

I've currently configured 4 dial-peers - two of them (Internal to External) are working fine, as described. But for the External to internal connection it does not work out. Troubleshooting for ccsip messages shows, that there are sip invites coming in on the voice gateway from the ISPs SIP trunk, but are not forwarded to CUCM.

Current Dial-Peers: (1 incoming from CUCM; 2 incoming from ISP; 10 outgoing to CUCM; 11 Outgoing to ISP)

dial-peer voice 1 voip
 description CUCM-VGW --- incoming ---
 session protocol sipv2
 incoming called-number 0T
 incoming uri via 2
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 2 voip
 description Versatel-VGW --- incoming ---
 session protocol sipv2
 incoming called-number 004930213089T
 incoming uri via 3
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 10 voip
 description VGW - CCM --- outgoing ---
 destination-pattern 004930213089T
 session protocol sipv2
 session target ipv4:10.5.42.200
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 11 voip
 description VGW - Versatel --- outgoing ---
 translation-profile outgoing POTS-OUT
 numbering-type unknown
 destination-pattern 0T
 session protocol sipv2
 session target sip-server
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

Looking forward to any advice in regards of this.

All incoming SIP Invites from the ISP Come in as

To: <004930213989XXX@172.17.17.4>

Kind Regards,

Chris

[Rajan]

Hi Chris,

As per the info given, this will match Dial peer 11 instead of 10 for outbound match. COuld you please share the below debugs for one failed call along with the calling and called number to check and confirm:

debug ccsip messages
debug voip ccapi inout

Thanks
Rajan

[bosechristoph]

Hello Rajan,

Thanks for helping out.

I fail to see how this is going to match Dial Peer 11 instead of Dial Peer 10.

Debug voip ccapi inout is not coming up with anything for the failed call (also - there is no signaling heard on the calling phone)

Debug ccsip messages output:

CUBE#debug ccsip message
SIP Call messages tracing is enabled
CUBE#
*Jan 22 11:21:55.895: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:004930213089100@172.17.17.4 SIP/2.0
Via: SIP/2.0/UDP 172.17.17.1:5060;branch=z9hG4bK55c07f6d9414e49b3
Max-Forwards: 70
From: <sip:443039111@172.17.17.1>;tag=712123059e
To: <sip:004930213089100@172.17.17.4>
Call-ID: 272b6dc8e40ca463
CSeq: 1190631592 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, REFER, REGISTER
Contact: <sip:443039111@172.17.17.1:5060;transport=udp>
P-Asserted-Identity: <sip:443039111@172.17.17.1>
Supported: replaces
User-Agent: Trinity 3.x M5T SIP Stack/4.2.14.18
Content-Type: application/sdp
Content-Length: 218

v=0
o=MxSIP 0 277 IN IP4 172.17.17.1
s=SIP Call
c=IN IP4 172.17.17.1
t=0 0
m=audio 6360 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

Obviously, these invites are coming in a couple of times :)

Regards,

Christoph

[Rajan]

Hi Christoph,

I said it will match dial peer 11 since you have mentioned 004930213989XXX in the previous message and not 004930213089XXX.

Based on the extension range you want to route to CUCM for example 910XXXX, you need to fine tune Dial peer 10 towards CUCM accordingly.

So after the invite message what do you see ? which side is disconnecting and with what cause code.

HTH
Rajan
Pls rate all useful posts

[bosechristoph]

Sorry for the wrong info, was a typo.

All I get is Invite messages for quite some time and at some point the call disconnects, I do not see any message regarding this on the voice gateway.

Well, I Want to forward the complete number (i.e. 004930213989100) towards CUCM and just can't wrap my head around the fact, that this is not working with the dial-peer provided.

Regards,

Chris

[Rajan · Accepted solution]

Hi Chris,

This is strange. I would suggest to check whether the SIP binding is proper and those IP addresses are allowed in the gateway toll fraud application as well as any firewalls.

SInce you dont see anything after the invite, you could probably try taking a capture on that interface to see whats happening exactly.

HTH
Rajan

[bosechristoph]

Thank you so much for the last reply. Toll Fraud prevention it was. Checking the trusted Networks and addresses showed that there was a typo in there, which did prevent the call routing.

Thanks a lot and have a great day,

Chris

[Rajan]

Glad its solved. Have a nice day and thanks for the rating :)