cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Announcements
209
Views
0
Helpful
9
Replies
Areyouserious
Beginner

CME - incoming call issues 400 - invalid host

Hi All,

I've been trying to figure out how to enable incoming calls on CME with a sip trunk.  Our CME is behind another border edge router, hence the ip route command in my config.

 

I'd like to have our DID number ring a group of our phones, but Can't figure out what commands I might have left out in order to route the in coming calls to the phones.

 

After days of scratching my head, I was wondering if anyone could help me where I might have gone wrong. 

 

If anyone could suggest any commands that ive left out (i'm sure perhaps a hunt group command?) i'd be most appreciative.  

 

Last time I recived any errors in the debug ccsip messages  was an invalid host message being sent by my CME to my sip trunk service when ever incoming calls would come in. 

 

Just not to certain about the voice translation profiles or dial peers.

Most of this config was provided by my voip provider, but not all was included, mainly how to route incoming calls.

 

Here is my config;;  - please note, all sensitive info has been replaced carefully with dummy info, but still remains consistent with config and logs in its original placement.  

 

 


Current configuration : 10478 bytes
!
! Last configuration change at 11:11:04 UTC Thu Mar 24 2022
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
!
!
crypto pki token default removal timeout 0
!
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
!
ip name-server 8.8.8.8
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice service voip
ip address trusted list

ipv4  202.36.52.125 255.255.255.0
ipv4 203.30.19.164 255.255.255.255
ipv4 203.32.124.160 255.255.255.224
ipv4 202.74.176.160 255.255.255.240
ipv4 202.74.176.176 255.255.255.248
ipv4 203.7.224.128 255.255.255.240
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 3600 min 3600
localhost dns:sip.simtex.com.au
no update-callerid
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
voice class sip-profiles 1000
request ANY sdp-header Connection-Info remove
response ANY sdp-header Connection-Info remove
!
!
!
!
voice translation-rule 1111
rule 1 /.*/ /612XXXXXXXX/ !Our DID here
!
voice translation-rule 1112
rule 1 /^0/ //
!
!
voice translation-profile Remove_0Prefix_ApplyCLID
translate calling 1111
translate called 1112
!
!
voice-card 0
!
!
!

!
redundancy
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
no ip address
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 192.168.2.5 255.255.255.0
duplex auto
speed auto
no shut
!
ip forward-protocol nd
no ip http server
no ip http secure-server
!
!
ip route 0.0.0.0 0.0.0.0 192.168.2.1
!
!
!
control-plane
!
!
voice-port 0/0/0
!
voice-port 0/0/1
!
!
!
mgcp profile default
!
!
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Demonstration peer from Simtex)
session protocol sipv2
session target ras
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2000 voip
description ** Outgoing 8-digit Local with 0 prefix (Demonstration peer from
translation-profile outgoing Remove_0Prefix_ApplyCLID
destination-pattern 0[2-9].......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2001 voip
description ** Outgoing 10-digit Local with 0 prefix (Demonstration peer fro
translation-profile outgoing Remove_0Prefix_ApplyCLID
destination-pattern 00[2-9]........
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2002 voip
description ** Outgoing 13 Local with 0 prefix (Demonstration peer from Simt
translation-profile outgoing Remove_0Prefix_ApplyCLID
destination-pattern 013[1-9]...
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2003 voip
description ** Outgoing 13/18 Local with 0 prefix (Demonstration peer from S
translation-profile outgoing Remove_0Prefix_ApplyCLID
destination-pattern 01[38]00......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
!
sip-ua
credentials username xxxxxxxx password 7 xxxxxxxx realm sip.simtex.com.au
keepalive target dns:sip.simtex.com.au
authentication username xxxxxxxx password 7 xxxxxxxx
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar dns:sip.simtex.com.au expires 1200
sip-server dns:sip.simtex.com.au
connection-reuse
host-registrar
g729-annexb override
!
!
!
telephony-service
max-ephones 40
max-dn 144 no-reg both
ip source-address 192.168.2.5 port 2000
max-conferences 8 gain -6
moh flash:music-on-hold.au
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1
number 2000 no-reg both

!
!
ephone-dn 2
number 2001 no-reg both

!
!
ephone-dn 3
number 2002 no-reg both

!
!
ephone-dn 4
number 2003 no-reg both

!
!
ephone-dn 5
number 2006 no-reg both

!
!
ephone 1
device-security-mode none
mac-address xxxx.xxxx.xxxx
button 1:1 3m2 4m3 5m4
button 6m5
!
!
!
ephone 2
device-security-mode none
mac-address xxxx.xxxx.xxxx
button 1:2 3m1 4m3 5m4
button 6m5
!
!
!
ephone 3
device-security-mode none
mac-address xxxx.xxxx.xxxx
button 1:3 3m1 4m2 5m4
button 6m5
!
!
!
ephone 4
device-security-mode none
mac-address xxxx.xxxx.xxxx
button 1:4 3m1 4m2 5m3
button 6m5
!
!
!
ephone 5
device-security-mode none
!
!
!
!

 

 

 


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

 

And here is an example of the debug ccsip messages being displayed when calling my DID from an outside number

 

 

 

 

 

Received:
INVITE sip:61285647585@202.36.52.125 SIP/2.0
Via: SIP/2.0/UDP 202.36.52.125:5060;branch=z9hG4bK515002d3;rport
Max-Forwards: 70
From: <sip:0485945256@sip.simtex.com.au>;tag=as7559ce03
To: <sip:61285647585@202.36.52.125>
Contact: <sip:0485945256@202.36.52.125:5060>
Call-ID: 25e5a4736050c133335a42cb61004581@sip.simtex.com.au
CSeq: 102 INVITE
User-Agent: SMTX_1610_0b0620
Date: Thu, 24 Mar 2022 15:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: rep
Router#laces, timer
P-Asserted-Identity: "0485945256" <sip:0485945256@sip.simtex.com.au>
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 1238438468 1238438468 IN IP4 202.36.52.125
s=Asterisk PBX 16.18.0
c=IN IP4 202.36.52.125
t=0 0
m=audio 10000 RTP/AVP 8 0 9 18 3 102 115 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:80
a=sendrecv

*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4B65F1B8) with key=[18] to table
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/ccsip_iwf_init:
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization: Entry...
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 202.36.52.125,Port 5060, Transport 1, SentBy Port 5060
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 192.168.2.5
*Mar 24 11:12:34.419: //-1/23D08EC1801B/SIP/State/sipSPIChangeState: 0x4B65F1B8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 202.36.52.125,Port 5060, Transport 1, SentBy Port 5060
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 202.36.52.125,Port 5060, Transport 1, SentBy Port 5060
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 192.168.2.5
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:100, category:100
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[-1], src[6]
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4B65F1B8 key=25e5a4736050c133335a42cb61004581@sip.simtex.com.au61285647585
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4B65F1B8 key=25e5a4736050c133335a42cb61004581@sip.simtex.com.au118140-21C7
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:400, container:49D22A78
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/Session-Timer/sipSTSLMain:
SE: 0;refresher:none peer refresher:none, flags:0, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 3 event
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPISendInviteResponse: Associated container=0x49D22A78 to Invite Response 400
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Transport/sipSPITransportSendMessage: msg=0x4CBD41BC, addr=202.36.52.125, port=5060, sentBy_port=5060, local_addr=, is_req=0, transport=1, switch=0, callBack=0x41C1F7D0
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Mar 24 11:12:34.427: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportLogicSendMsg: connection-reuse configured, listen conn-id : 0
*Mar 24 11:12:34.427: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4CBD41BC, addr=202.36.52.125, port=5060, local_addr=, connId=0 for UDP
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/State/sipSPIChangeState: 0x4B65F1B8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Mar 24 11:12:34.431: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 202.36.52.125:5060;branch=z9hG4bK515002d3;rport
From: <sip:0485945256@sip.simtex.com.au>;tag=as7559ce03
To: <sip:61285647585@202.36.52.125>;tag=118140-21C7
Date: Thu, 24 Mar 2022 11:12:34 GMT
Call-ID: 25e5a4736050c133335a42cb61004581@sip.simtex.com.au
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

1 ACCEPTED SOLUTION

Accepted Solutions
Roger Kallberg
VIP Mentor

Suggest that you match on the via header for the inbound direction from your service provider. For more information on this please see this document.

Apart from this I would recommend to remove these lines from this dial peer.

dial-peer voice 1000 voip
 no permission term
 no session target ras
 no incoming called-number .%

The last line is replaced by the match on via header as suggested above.



Response Signature


View solution in original post

9 REPLIES 9
Roger Kallberg
VIP Mentor

Suggest that you match on the via header for the inbound direction from your service provider. For more information on this please see this document.

Apart from this I would recommend to remove these lines from this dial peer.

dial-peer voice 1000 voip
 no permission term
 no session target ras
 no incoming called-number .%

The last line is replaced by the match on via header as suggested above.



Response Signature


Hey Roger,

 

Thankyou so much for your help.

 

As you mentioned, I think it may have been these commands that may have been causing an issue, so I added the reverse of them with the (no) infront;
dial-peer voice 1000 voip
 no permission term
 no session target ras
 no incoming called-number .%

 

Phone calls are now coming in to a hunt list I made:

 

voice hunt-group 1 parallel
list 2000,2002,2003,2004                     
pilot (FULL DID NUMBER IN HERE including country code)

 

The system seems to be happy working behind another NAT router, and despite the SIP invites being addressed to our WAN ip, it's still working. I guess maybe it might not have been a header issue after all? 

 

Other than that, not to sure what happened????

 

Thanks again

Areyouserious
Beginner

Sorry forgot to post the output

 

 

 

 

Received:
INVITE sip:61285647585@202.36.52.125 SIP/2.0
Via: SIP/2.0/UDP 202.36.52.125:5060;branch=z9hG4bK515002d3;rport
Max-Forwards: 70
From: <sip:0485945256@sip.simtex.com.au>;tag=as7559ce03
To: <sip:61285647585@202.36.52.125>
Contact: <sip:0485945256@202.36.52.125:5060>
Call-ID: 25e5a4736050c133335a42cb61004581@sip.simtex.com.au
CSeq: 102 INVITE
User-Agent: SMTX_1610_0b0620
Date: Thu, 24 Mar 2022 15:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: rep
Router#laces, timer
P-Asserted-Identity: "0485945256" <sip:0485945256@sip.simtex.com.au>
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 1238438468 1238438468 IN IP4 202.36.52.125
s=Asterisk PBX 16.18.0
c=IN IP4 202.36.52.125
t=0 0
m=audio 10000 RTP/AVP 8 0 9 18 3 102 115 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:80
a=sendrecv

*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4B65F1B8) with key=[18] to table
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/ccsip_iwf_init:
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
*Mar 24 11:12:34.419: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization: Entry...
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 202.36.52.125,Port 5060, Transport 1, SentBy Port 5060
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 192.168.2.5
*Mar 24 11:12:34.419: //-1/23D08EC1801B/SIP/State/sipSPIChangeState: 0x4B65F1B8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 202.36.52.125,Port 5060, Transport 1, SentBy Port 5060
*Mar 24 11:12:34.419: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 202.36.52.125,Port 5060, Transport 1, SentBy Port 5060
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 192.168.2.5
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:100, category:100
*Mar 24 11:12:34.423: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[-1], src[6]
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4B65F1B8 key=25e5a4736050c133335a42cb61004581@sip.simtex.com.au61285647585
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
*Mar 24 11:12:34.423: //-1/23D08EC1801B/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4B65F1B8 key=25e5a4736050c133335a42cb61004581@sip.simtex.com.au118140-21C7
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:400, container:49D22A78
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/Session-Timer/sipSTSLMain:
SE: 0;refresher:none peer refresher:none, flags:0, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 3 event
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sipSPISendInviteResponse: Associated container=0x49D22A78 to Invite Response 400
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Transport/sipSPITransportSendMessage: msg=0x4CBD41BC, addr=202.36.52.125, port=5060, sentBy_port=5060, local_addr=, is_req=0, transport=1, switch=0, callBack=0x41C1F7D0
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Mar 24 11:12:34.427: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportLogicSendMsg: connection-reuse configured, listen conn-id : 0
*Mar 24 11:12:34.427: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4CBD41BC, addr=202.36.52.125, port=5060, local_addr=, connId=0 for UDP
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response
*Mar 24 11:12:34.427: //-1/23D08EC1801B/SIP/State/sipSPIChangeState: 0x4B65F1B8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Mar 24 11:12:34.431: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 202.36.52.125:5060;branch=z9hG4bK515002d3;rport
From: <sip:0485945256@sip.simtex.com.au>;tag=as7559ce03
To: <sip:61285647585@202.36.52.125>;tag=118140-21C7
Date: Thu, 24 Mar 2022 11:12:34 GMT
Call-ID: 25e5a4736050c133335a42cb61004581@sip.simtex.com.au
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Roger Kallberg
VIP Mentor

It looks like the call is coming from an IP, 202.36.52.125, that you don’t have in your trust list.

ip address trusted list
ipv4 203.30.19.164 255.255.255.255
ipv4 203.32.124.160 255.255.255.224
ipv4 202.74.176.160 255.255.255.240
ipv4 202.74.176.176 255.255.255.248
ipv4 203.7.224.128 255.255.255.240

This would trigger the toll bypass protection in the gateway to decline the call. It works in the way that anything specifically set on dial peer(s) as the destination is automatically trusted. Anything else has to be added to the trust list.

Apart from this, as you said that there is another SBC in between the CME and the service provider do you really need to have any registration configuration in this box? Wouldn’t that be handled by the other router that acts as the SBC?



Response Signature


Hi Roger thanks for your replies, 

 

Sorry for the confusion,  The IP required is listed there, it's just I replaced the ip address in the logs with an alias to hide the real one from the public viewing these posts. 

 

This should be all corrected now.

 

As for the setup,  I simply have a CME ISR router 2811 setup behind an existing Cisco router that is not end of life just yet, as placing the now end of life 2811 on the border edge could be risky from what I can gather.   I'm trying to set up this system to learn off of it, but also make some calls when needed.   

 

Just also curious, if I were to get this working, which telephones would ring without a hunt group configured for my incoming DID?

 

Also with the sip invite packets for incoming calls from my SIP provider,  do these need to match an IP address that the CME router is familiar with such as a loopback?   I think these invite packets are directed at my public WAN ip known as (202.36.52.125)? so if I have this correct,  these  invites need to be directed at maybe.....say a loopback on the CME???

 

 

What do your other router do, is it setup as a simple L3 device or is it acting as an session border controller. If you have the edge router acting as an SBC it should be setup to not pass along the information from the service provider to upstream devices.

On your question about the IP, as I wrote in my response before the router needs to know about the IP that the invite comes from, otherwise it will decline the call.

What would ring is depending on the called number inbound from the service provider. A phone with that called number set as it’s directory number will ring.



Response Signature


Hey Roger thanks again for your reply,

 

So basically what I’m doing is I have a network termination unit modem for our Internet service and then that is bridged into a Cisco router that I have with Nat translation switched on between the wide area network, and in the internal network, the 192.168.2.0 network  is branching off the border router, (cme 2811) and there’s a static route between those particular two Routers. CBAC  is also turned on the boarder router.  So obviously if I’m guessing this correctly there would’ve been no problem if the CME router was in place of the border edge router that I’m using because it would hold the public WAN address of which my sip provider is forwarding the SIP invites???

 

Thanks mate, appreciate your time and efforts

The issue is the the border router with your setup will not modify the content inside the SIP signaling, so the router on the inside of the NAT router still “sees” the public IP information and that is why the call fails. Look at the second link in my other response for how this could be mitigated.



Response Signature


Maybe either of these link1 link2 could help.



Response Signature


Create
Recognize Your Peers
Content for Community-Ad