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SIP/2.0 400 Bad Request - 'Invalid IP Address' & 'Invalid IP Host'

Joe_Tseng1
Level 1
Level 1

I use Webex Calling and integrate CSR 1000v(Azure) as Local Gateway

Webex Calling > CSRv1000 > ITSP

CSRv1000 / privacy ip-10.0.0.4 / public ip-20.59.121.X

CSRv1000 Public IP:20.59.121.172 / CSRv1000 privacy IP:10.0.0.4
ITSP IP:211.75.165.204

Webex Calling uses CSR 1000v as Local gateway and integrates PSTN

It works fine when using Webex Calling to dial PSTN. But there will be problems when dialing PSTN to enter Webex Calling.

A message of SIP/2.0 400 Bad Request - 'Invalid IP Host' will appear.

How to fix this issue? thx a lot.


---SIP Options---

*May 30 02:43:02.448: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:20.59.121.172 SIP/2.0
Via: SIP/2.0/UDP 211.75.165.204:5060;branch=z9hG4bK77b256f0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@211.75.165.204>;tag=as77209e78
To: <sip:20.59.121.172>
Contact: <sip:asterisk@211.75.165.204:5060>
Call-ID: 2702203524f45fe64eff1fc076a45dc5@211.75.165.204:5060
CSeq: 102 OPTIONS
User-Agent: cs210
Date: Tue, 30 May 2023 02:43:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, UPDATE, PRACK
Supported: replaces, timer
Content-Length: 0


*May 30 02:43:02.449: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 211.75.165.204:5060;branch=z9hG4bK77b256f0
From: "asterisk" <sip:asterisk@211.75.165.204>;tag=as77209e78
To: <sip:20.59.121.172>;tag=CBFE213-16C2
Date: Tue, 30 May 2023 02:43:02 GMT
Call-ID: 2702203524f45fe64eff1fc076a45dc5@211.75.165.204:5060
Server: Cisco-SIPGateway/IOS-17.3.3
CSeq: 102 OPTIONS
Content-Length: 0


---SIP Call---
Calling Number:0958971125
Called Number :+886289797705

 

82794: *May 30 02:27:47.265: //4294967295/65D491F3A588/
------------------ Cover Buffer ---------------
Search-key = 0958971125:+886289797705:NA
Timestamp = *May 30 02:27:46.206
CallID = NA
Peer-CallID = 0
Correlator = NA
Called-Number = +886289797705
Calling-Number = 0958971125
SIP CallID = 7f977c100621c55d5b0b0d4b494260bd@211.75.165.204:5060
SIP SessionID =
GUID = 65D491F3A588
-----------------------------------------------
82770: *May 30 02:27:46.206: //4294967295/65D491F3A588/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+886289797705@20.59.121.172:5060 SIP/2.0
Via: SIP/2.0/UDP 211.75.165.204:5060;branch=z9hG4bK7ec0b713
Max-Forwards: 70
From: "0958971125" <sip:0958971125@211.75.165.204>;tag=as1c1ac4e6
To: <sip:+886289797705@20.59.121.172:5060>
Contact: <sip:0958971125@211.75.165.204:5060>
Call-ID: 7f977c100621c55d5b0b0d4b494260bd@211.75.165.204:5060
CSeq: 102 INVITE
User-Agent: cs210
Date: Tue, 30 May 2023 02:27:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, UPDATE, PRACK
Supported: replaces, timer
P-Asserted-Identity: "0958971125" <sip:0958971125@211.75.165.204>
Content-Type: application/sdp
Content-Length: 297

v=0
o=root 62025466 62025466 IN IP4 211.75.165.204
s=Asterisk PBX 11.6-cert6
c=IN IP4 211.75.165.204
t=0 0
m=audio 10194 RTP/AVP 8 0 100
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=sendrecv

82772: *May 30 02:27:46.207: //4294967295/65D491F3A588/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_NONE, Next State = STATE_IDLE, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
82773: *May 30 02:27:46.208: //4294967295/65D491F3A588/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir:Inbound, Peer-Tag: 100
82774: *May 30 02:27:46.208: //4294967295/65D491F3A588/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x1700ADA, Originated at:0x1700A43, Cause Code = 100
82775: *May 30 02:27:46.208: //4294967295/65D491F3A588/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_IDLE, Next State = STATE_DISCONNECTING, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
82776: *May 30 02:27:46.208: //4294967295/65D491F3A588/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 211.75.165.204:5060;branch=z9hG4bK7ec0b713
From: "0958971125" <sip:0958971125@211.75.165.204>;tag=as1c1ac4e6
To: <sip:+886289797705@20.59.121.172:5060>;tag=CB1E702-245E
Date: Tue, 30 May 2023 02:27:46 GMT
Call-ID: 7f977c100621c55d5b0b0d4b494260bd@211.75.165.204:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-17.3.3
Session-ID: 23ce669123a456cdb99608ddcd555ec7;remote=b1296be3a963597fbd4c3a0fcf2d69eb
Content-Length: 0

3 Replies 3

b.winter
VIP
VIP

Have you followed the Cisco configuration guide on how to configure a CUBE for local GW?
Can you post the config and a full debug of a non-working call?
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

Edit:
Probably what you are missing is SIP-profiles, that change the public IP to the CUBE's private IP.

The INVITE from the provider is sent to the public IP of the CUBE, the Azure FW NATs the public IP to the private IP only in the IP-header, but in the SIP-header, there are still the public IPs.
And as CUBE only knows its private IP, it doesn't feel responsible for those requests.

Joe_Tseng1
Level 1
Level 1

Hi b.winter , thanks for your reply. i've config sip-profiles 100 and assign to incoming dial-peer but still not work. do you have standard config?

 

 

For the connection between the CUBE and Webex there are docs like this: https://help.webex.com/en-us/article/jr1i3r/Configure-Local-Gateway-on-Cisco-IOS-XE-for-Webex-Calling#id_100573

But I don't have any "standard config" for the connection between CUBE and your provider. As you can imagine, nobody will have every config for every provider in the world.

About your config:
Your SIP profile 100 is wrong. Your are new changing the private IP to the public IP. But you want to have it the other way round.
The message is coming in with public IP, so you want to change that to the private IP of the CUBE.

You also need:

voice service voip
 sip
  sip-profiles inbound
!
dial-peer voice 100
 sip-profiles 100 inbound

And since you want to manipulate the SIP message in the inbound direction, you need to set the sip profile for working inbound.
Per default, they only work in outbound direction.