CUBE SIP 403 Request From Unknown Source

[uastac2011]

Hi Guys,

Need some help. We are implementing SIP connection and I am new for this. I cannot make it work. I already did what the article or forums but still failed.

Here is the topology. Attached files are sip config and debug. 
I am hoping that you can help me. Thank you so much.

[uastac2011]

Hi Vaijanath 

attached files are the debug. thank you.

[Vaijanath Sonvane]

Please upload your current configuration. Also, for your incoming debug I do not see any incoming call related messages.

[uastac2011]

Hi Vaijanath

Here is what they gave to us.

Please attached debugs and config.

[TONY SMITH]

OK based on that information it looks like you need to be sending and receiving your SIP signalling on 192.168.1.202.  So that address need to be configured on and interface of CUBE and set as the bind address for both signalling and media.

I had a quick look at your config and I'd suggest matching incoming dial peer by URI rather than by called number.  And I note you have session target of "sip server" used for both CUCM and ITSP dial peers.  

[uastac2011]

Hi Tony
yesterday i coordinated with telco. they did a trace call and saw the reject coming from them, they will check why their sbc is rejecting our sip.

can you help me how to configure sip uri.

thank you so much.

[Gregory Brunn]

So we are coming back around to the carrier is denying the call?

[uastac2011]

Yes sir

[TONY SMITH]

Back where we started really.   Some progress was seen on the 19th when at least an incoming call was being received.  If we could have seen configuration and debugs we might have been able to move things on. Just off the top of my head it looked like a dial peer issue in the CUBE which could have been sorted out.   I don't really understand why the configuration was changed after that.

[uastac2011]

Hi Tony
the only changed i did was to put the 192.168.1.202 to cucm.

[TONY SMITH]

OK, so from the debugs on the 19th we can see you were nearer to a working system when using 192.168.1.202 as the CUBE's address for signalling to the ITSP.  The 404 error that you saw with incoming call could have been dial peer issue, not really possible to be clear as we didn't see full debugs or a configuration.  

So two options come to mind.  (1) Revert to that configuration

Or (2) ask your ITSP to accept 10.229.141.222 as your signalling address.  That would be more normal, an ITSP doesn't usually see any further into a customer's network than the WAN interface.

Which ever you go for, please post up debugs (debug ccsip mess) showing both call legs, for inbound and outbound call attempts.  Also post up your CUBE configuration

[uastac2011]

Incoming is now working.
changed the destination address to sip provider.
but still no outgoing call.
we are having error 487.

00033667.001 |18:46:13.248 |AppInfo |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 10.251.48.148:[5060]:
[2253,NET]
INVITE sip:95324050@10.251.48.148:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK10b3265811a
From: "8918330" <sip:028918330@192.168.1.202>;tag=710~877633c6-2d58-42d6-9ad3-513696b0e924-27135969
To: <sip:95324050@10.251.48.148>
Date: Thu, 29 Aug 2019 10:46:13 GMT
Call-ID: 36ef4e00-d671acf5-fa-ca01a8c0@192.168.1.202
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 0921652736-0000065536-0000000026-3389106368
Session-Expires: 1800
P-Asserted-Identity: "8918330" <sip:028918330@192.168.1.202>
Remote-Party-ID: "8918330" <sip:028918330@192.168.1.202>;party=calling;screen=yes;privacy=off
Contact: <sip:028918330@192.168.1.202:5060>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 201

v=0
o=CiscoSystemsCCM-SIP 710 1 IN IP4 192.168.1.202
s=SIP Call
c=IN IP4 192.168.1.202
t=0 0
m=audio 24666 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Trying from Provider:

00033668.001 |18:46:13.253 |AppInfo |//SIP/SIPUdp/wait_SdlDataInd: Incoming SIP UDP message size 311 from 10.251.48.148:[5060]:
[2254,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK10b3265811a
From: "8918330" <sip:028918330@192.168.1.202>;tag=710~877633c6-2d58-42d6-9ad3-513696b0e924-27135969
To: <sip:95324050@10.251.48.148>
Call-ID: 36ef4e00-d671acf5-fa-ca01a8c0@192.168.1.202
CSeq: 101 INVITE
Content-Length: 0

Now we got 487 error code from provider.

00033684.001 |18:46:13.368 |AppInfo |//SIP/SIPUdp/wait_SdlDataInd: Incoming SIP UDP message size 461 from 10.251.48.148:[5060]:
[2258,NET]
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK10b3265811a
From: "8918330" <sip:028918330@192.168.1.202>;tag=710~877633c6-2d58-42d6-9ad3-513696b0e924-27135969
To: <sip:95324050@10.251.48.148>;tag=jmpesmkf
Call-ID: 36ef4e00-d671acf5-fa-ca01a8c0@192.168.1.202
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

[TONY SMITH]

Can you show us debugs for the successful incoming call?  Just want to see how the ITSP sets some parameters.  

[uastac2011]

Hi Tony

Attached files are the rtmt logs. There is no logs in the cube i think it is because the destination address in sip trunk is the scb ip  and it was already bypass the cube.

[uastac2011]

outgoing call is now working. i stripped the 9 (5324050)