Sip Trunk to ISP on CME.

Rajan R · ‎09-15-2022

Hello All,

I have a CME which registers my SIP Phones for internal calls.

We have a SIP Trunk delivered by ISP now with only the username, ISP says no password is required.

ISP says this should be configured as a Peer Trunk. How do i achieve this on the CME ?

Thanks a lot.

Roger Kallberg · ‎09-15-2022

Likely you’ll only create an inbound and outbound dial dial towards your service provider and make sure that you put the correct bind statements for the outside interface for your SBC (CME GW) on them.

Rajan R · ‎09-15-2022

Thanks for the response. FYI, the SIP connection is also terminating on the same CME Router. My LAN is on 0/0/0 and the SIP connection from ISP is on 0/0/1.

Where do I use these bind statements ?

Thanks

b.winter · ‎09-15-2022

On the dial-peer as @Roger Kallberg mentioned

Rajan R · ‎09-15-2022

Thanks. I tired doing that. This is what I see.My gig0/0/0 is my LAN and gig 0/0/1 is towards my isp which has the Ip 192.168.2.2. In the below log, why is the "To:" column going to my LAN instead of my 0/0/1 interface. Where am I going wrong ? Tried doing an outgoing call from my Cisco phone to a cell phone

2732: *Sep 15 13:14:59.855: //271/3C52A3E3822A/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.63:5060;branch=z9hG4bK42abc6aa
From: "XXXXX8900" <sip:XXXXX8900@192.168.1.5>;tag=ec01d50bf0af00246b703191-49e24ebd
To: <sip:XXXXXX091@192.168.1.5>;tag=622207-C79
Date: Thu, 15 Sep 2022 13:14:59 GMT
Call-ID: ec01d50b-f0af001e-615050aa-01343fa5@192.168.1.63
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-17.3.4a
Reason: Q.850;cause=3
Session-ID: 1009475100105000a000ec01d50bf0af;remote=b79c5ca33a435c43965be743f8071a0f
Content-Length: 0

b.winter · ‎09-15-2022

Hi, nobody can help you, if you just post small pieces of a debug.

Post the following debug output and the config (without any sensitive data like username / password / secrets...):
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

Rajan R · ‎09-15-2022

Please find below..

Cisco-4321#
*Sep 15 13:37:02.969: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:XXX3789XX@192.168.2.2;user=phone SIP/2.0
Accept: application/sdp,application/dtmf-relay
Allow: PRACK,ACK,CANCEL,BYE,SUBSCRIBE,NOTIFY,INVITE,REFER,OPTIONS,PUBLISH,INFO,UPDATE,REGISTER
Allow-Events: hold,talk
Call-ID: OA6AB51491XXX2165XXX90DC72C6
Contact: <sip:XXX2165XXX@192.168.2.1;user=phone>
Content-Type: application/sdp
CSeq: 324 INVITE
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
Max-Forwards: 70
P-Asserted-Identity: <sip:XXX2165XXX@192.168.2.1;user=phone>
Privacy: none
Supported: replaces
To: <sip:XXX3789XX@192.168.2.2;user=phone>
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
Content-Length: 158

v=0
o=User1 3872230564 3872230564 IN IP4 192.168.2.1
s=Session SDP
c=IN IP4 192.168.2.1
t=0 0
m=audio 16416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20

*Sep 15 13:37:02.972: //-1/50F66C608273/CCAPI/cc_api_call_setup_ind_common:
Interface=0x7FA893509D38, Call Info(
Calling Number=XXX2165XXX,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=XXX3789XX(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=2, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=325
*Sep 15 13:37:02.974: //325/50F66C608273/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
To: <sip:XXX3789XX@192.168.2.2;user=phone>
Date: Thu, 15 Sep 2022 13:37:02 GMT
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-17.3.4a
Session-ID: 00000000000000000000000000000000;remote=5668dfd7e5cf5900840dac98116258d5
Content-Length: 0

*Sep 15 13:37:02.975: //325/50F66C608273/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x7FA893509D38, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=XXX2165XXX,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=XXX3789XX(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=40001, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Sep 15 13:37:02.977: //325/50F66C608273/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
To: <sip:XXX3789XX@192.168.2.2;user=phone>;tag=765279-9E0
Date: Thu, 15 Sep 2022 13:37:02 GMT
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.1.5 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-17.3.4a
Reason: Q.850;cause=47
Session-ID: 00000000000000000000000000000000;remote=5668dfd7e5cf5900840dac98116258d5
Content-Length: 0

*Sep 15 13:37:02.983: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:XXX3789XX@192.168.2.2;user=phone SIP/2.0
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 ACK
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
Max-Forwards: 70
To: <sip:XXX3789XX@192.168.2.2;user=phone>;tag=765279-9E0
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
Content-Length: 0

3010: *Sep 15 13:37:03.550: //4294967295/xxxxxxxxxxxx/
------------------ Cover Buffer ---------------
Search-key = ::NA
Timestamp = *Sep 15 13:37:02.974
CallID = NA
Peer-CallID = 325
Correlator = NA
Called-Number =
Calling-Number =
SIP CallID =
SIP SessionID =
GUID =
-----------------------------------------------
2971: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir:Outbound, Peer-Tag: 40001
2972: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/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
2973: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/FSM/IWF: Event = E_SIP_IWF_EV_SET_MODE, Current State = CNFSM_CONTAINER_STATE, Next State = CNFSM_NO_STATE_CHANGE
2974: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/FSM/IWF: Event = E_SIP_IWF_EV_PRE_SETUP, Current State = S_SIP_IWF_SDP_IDLE, Next State = CNFSM_NO_STATE_CHANGE
2980: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/FSM/IWF: Event = E_SIP_IWF_EV_PEER_MULTIMEDIA_CHANNEL_IND, Current State = S_SIP_IWF_SDP_IDLE, Next State = CNFSM_NO_STATE_CHANGE
2981: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Media Stream Parameters: Stream Type = voice-only, Stream State = STREAM_ADDING Negotiated Codec = No Codec , Negotiated DTMF Type = inband-voice, Stream Index = 1
2982: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/API: voip_rtp_allocate_port (8144)
2983: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/FSM/Media-State: Event = E_IPIP_MEDIA_SERV_EV_PEER_CHNL_IND, Current State = S_IPIP_MEDIA_SERV_STATE_IDLE, Next State = S_IPIP_MEDIA_SERV_STATE_INIT_XCODER_RESERVED
2984: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Software-Error: Type: Error in Transcoder Reservation, Code-location:0x36005A1
2985: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Error: sipSPICodecTranscoder: Disjoint set & xcoder reservation failed. Disconnect call
2986: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x3003DE8, Originated at:0x3003DC7, Cause Code = 47
2987: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/API: cc_api_call_disconnected (0)
2988: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/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
2989: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Error: sipSPI_sip_BWCAC_calc_max_audio_bw: could not find max bw codec!!!!
2990: *Sep 15 13:37:02.974: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x260070A, Originated at:0x260070B, Cause Code = 47
2991: *Sep 15 13:37:02.975: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_CC_CALL_DISCONNECT, Current State = STATE_DISCONNECTING
2992: *Sep 15 13:37:02.976: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/API: cc_api_call_disconnect_done (0)
2993: *Sep 15 13:37:02.976: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_DISCONNECTING, Next State = STATE_DEAD, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
2994: *Sep 15 13:37:02.976: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/MISC/Error: sipSPIFlushDeferredQueue: Invalid deferredQueue
2995: *Sep 15 13:37:02.976: //4294967295/xxxxxxxxxxxx/CUBE_VT/SIP/API: voip_rtp_release_port (8144)
3011: *Sep 15 13:37:03.551: //325/50F66C608273/
------------------ Cover Buffer ---------------
Search-key = XXX2165XXX:XXX3789XX:325
Timestamp = *Sep 15 13:37:02.969
CallID = 325
Peer-CallID = 326
Correlator = NA
Called-Number = XXX3789XX
Calling-Number = XXX2165XXX
SIP CallID = OA6AB51491XXX2165XXX90DC72C6
SIP SessionID =
GUID = 50F66C608273
-----------------------------------------------
2958: *Sep 15 13:37:02.969: //325/50F66C608273/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:XXX3789XX@192.168.2.2;user=phone SIP/2.0
Accept: application/sdp,application/dtmf-relay
Allow: PRACK,ACK,CANCEL,BYE,SUBSCRIBE,NOTIFY,INVITE,REFER,OPTIONS,PUBLISH,INFO,UPDATE,REGISTER
Allow-Events: hold,talk
Call-ID: OA6AB51491XXX2165XXX90DC72C6
Contact: <sip:XXX2165XXX@192.168.2.1;user=phone>
Content-Type: application/sdp
CSeq: 324 INVITE
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
Max-Forwards: 70
P-Asserted-Identity: <sip:XXX2165XXX@192.168.2.1;user=phone>
Privacy: none
Supported: replaces
To: <sip:XXX3789XX@192.168.2.2;user=phone>
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
Content-Length: 158

v=0
o=User1 3872230564 3872230564 IN IP4 192.168.2.1
s=Session SDP
c=IN IP4 192.168.2.1
t=0 0
m=audio 16416 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20

2960: *Sep 15 13:37:02.969: //325/50F66C608273/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
2961: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir:Inbound, Peer-Tag: 2
2962: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/FSM/Offer-Answer: Event = E_SIP_INVITE_SDP_RCVD, Current State = S_SIP_EARLY_DIALOG_IDLE, Next State = S_SIP_EARLY_DIALOG_OFFER_RCVD
2963: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/FSM/IWF: Event = E_SIP_IWF_EV_RCVD_SDP, Current State = S_SIP_IWF_SDP_IDLE, Next State = S_SIP_IWF_SDP_RCVD_AWAIT_PEER_EVENT
2964: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/MISC/Media Stream Parameters: Stream Type = voice-only, Stream State = STREAM_ADDING Negotiated Codec = g711alaw, Negotiated DTMF Type = inband-voice, Stream Index = 1
2965: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/API: cc_api_update_interface_cac_resource (0)
2966: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/API: voip_rtp_allocate_port (8142)
2967: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/MISC/Media Stream Parameters: Stream Type = voice-only, Stream State = STREAM_ADDING Negotiated Codec = g711alaw, Negotiated DTMF Type = inband-voice, Stream Index = 1
2968: *Sep 15 13:37:02.970: //325/50F66C608273/CUBE_VT/SIP/API: cc_api_call_setup_ind_with_callID (0)
2969: *Sep 15 13:37:02.971: //325/50F66C608273/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_IDLE, Next State = STATE_RECD_INVITE, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
2970: *Sep 15 13:37:02.973: //325/50F66C608273/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
To: <sip:XXX3789XX@192.168.2.2;user=phone>
Date: Thu, 15 Sep 2022 13:37:02 GMT
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-17.3.4a
Session-ID: 00000000000000000000000000000000;remote=5668dfd7e5cf5900840dac98116258d5
Content-Length: 0

2975: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/FSM/IWF: Event = E_SIP_IWF_EV_SET_MODE, Current State = CNFSM_CONTAINER_STATE, Next State = CNFSM_NO_STATE_CHANGE
2976: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/API: voip_rtp_create_session (0)
2977: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/API: voip_rtp_set_non_rtp_call (0)
2978: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/API: voip_rtp_update_callinfo (0)
2979: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_CC_CALL_PROCEEDING, Current State = STATE_RECD_INVITE
2996: *Sep 15 13:37:02.976: //325/50F66C608273/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x260070A, Originated at:0x260070B, Cause Code = 47
2997: *Sep 15 13:37:02.976: //325/50F66C608273/CUBE_VT/SIP/API: cc_api_update_interface_cac_resource (0)
2998: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_CC_CALL_DISCONNECT, Current State = STATE_RECD_INVITE
2999: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_RECD_INVITE, Next State = STATE_DISCONNECTING, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
3000: *Sep 15 13:37:02.975: //325/50F66C608273/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_DISCONNECTING, Next State = STATE_DISCONNECTING, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
3001: *Sep 15 13:37:02.976: //325/50F66C608273/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
To: <sip:XXX3789XX@192.168.2.2;user=phone>;tag=765279-9E0
Date: Thu, 15 Sep 2022 13:37:02 GMT
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.1.5 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-17.3.4a
Reason: Q.850;cause=47
Session-ID: 00000000000000000000000000000000;remote=5668dfd7e5cf5900840dac98116258d5
Content-Length: 0

3003: *Sep 15 13:37:02.982: //325/50F66C608273/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:XXX3789XX@192.168.2.2;user=phone SIP/2.0
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 ACK
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
Max-Forwards: 70
To: <sip:XXX3789XX@192.168.2.2;user=phone>;tag=765279-9E0
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
Content-Length: 0

3004: *Sep 15 13:37:02.983: //325/50F66C608273/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_NEW_MESSAGE, Current State = STATE_DISCONNECTING
3005: *Sep 15 13:37:02.983: //325/50F66C608273/CUBE_VT/SIP/API: voip_rtp_delete_dp_session (0)
3006: *Sep 15 13:37:02.984: //325/50F66C608273/CUBE_VT/SIP/API: cc_api_call_disconnect_done (0)
3007: *Sep 15 13:37:02.984: //325/50F66C608273/CUBE_VT/SIP/FSM/SPI-State-Change: Current State = STATE_DISCONNECTING, Next State = STATE_DEAD, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE
3008: *Sep 15 13:37:02.984: //325/50F66C608273/CUBE_VT/SIP/MISC/Error: sipSPIFlushDeferredQueue: Invalid deferredQueue
3009: *Sep 15 13:37:02.984: //325/50F66C608273/CUBE_VT/SIP/API: voip_rtp_release_port (8142)
Cisco-4321#
Cisco-4321#
Cisco-4321#
Cisco-4321#sh run
Cisco-4321#sh running-config
Building configuration...

Current configuration : 9372 bytes

version 17.3
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
service call-home
platform qfp utilization monitor load 80
platform punt-keepalive disable-kernel-core
!
hostname Cisco-4321
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
--More-- !
address-family ipv6
exit-address-family
!
!
no aaa new-model

!
!
!
!
!
!
!
!
login on-success log
!
!
!
!
!
--More-- !
!
subscriber templating
multilink bundle-name authenticated
no device-tracking logging theft
!
!
!
!
!
!
!
crypto pki trustpoint TP-self-signed-3535921534
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3535921534
revocation-check none
rsakeypair TP-self-signed-3535921534
!
crypto pki trustpoint SLA-TrustPoint
enrollment pkcs12
revocation-check crl
!
!
--More-- crypto pki certificate chain TP-self-signed-3535921534
certificate self-signed 01
30820330 30820218 A0030201 02020101 300D0609 2A864886 F70D0101 05050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 33353335 39323135 3334301E 170D3232 30393036 30383430
31395A17 0D333230 39303530 38343031 395A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D33 35333539
32313533 34308201 22300D06 092A8648 86F70D01 01010500 0382010F 00308201
0A028201 0100A12D 7C9C879D 87C0945F 43B9C670 A1C6624F 6C57A998 1CBE9906
13718B73 137C6276 6F54C3DE EBDBD698 EE45DDFD 4B37B692 BE741D68 C3F46F30
7F017FF6 FB83F434 02FFBAC6 F65B3292 E5B12DE0 B9E535CB 3BC333AF EA067783
2DE5E2F4 A1C80752 3C1302D2 CC7759FC EE57E8F9 8486F019 6734C8DC 9A023000
71222360 386F2C8E 947F363E 1ADB3E45 8A2F814F ACF12132 8573395C EC8C9A21
671D61D1 39C4EBFF 463D4200 82D4019D A628DFD9 9E41A552 59BCAAE3 1AC69319
7322FFBA DC7E34DA 6FE6BF3C B41A4864 9038A5AA A09DC65E 382AAC73 415C0121
A261CC61 D9D2C287 F9B588B9 EA2C900D 82681D4E 2F8922B0 AE1F3589 09A5373D
964E9271 F5FD0203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF
301F0603 551D2304 18301680 145C6132 C438F26E 8ECD6CC8 75127CB3 F1768AE2
A8301D06 03551D0E 04160414 5C6132C4 38F26E8E CD6CC875 127CB3F1 768AE2A8
300D0609 2A864886 F70D0101 05050003 82010100 000FAC7A 4155C5F0 81D825BF
9CD65847 8C36C38A 5DAF5B40 A627C288 4A6804AD 0E0D3596 E9B2FABE 6D0669D7
C8D138EE 7E392E93 2C917EDD F9D77B5B C8763B29 47B7C07B B503D613 5EE14B19
35AB391D 73E7AAFB F32F03C0 5F890212 B1DEB522 9245E889 84B0FD24 4D9BD716
--More-- 409EDB74 4D62932B F6A6E76E 2A436EC6 E8B74B79 AA47C386 C247032B CCDD5682
FFFF010F 2EA43AD1 39997676 56716D30 A88D2E9E 6D4E124F B41CEFF2 779C91EE
C37DFFB5 FBA26DB6 271F2398 46FC2F59 EC1E54D3 B6DB831A 4D69D082 5B6DE7CB
76E418F1 E448DDD7 C56D917D A3142569 84675DCC AAE55523 B97F7013 541A8488
7BFC12FF 5ACE3710 797342F1 45FFB2F8 92141DCE
quit
crypto pki certificate chain SLA-TrustPoint
certificate ca 01
30820321 30820209 A0030201 02020101 300D0609 2A864886 F70D0101 0B050030
32310E30 0C060355 040A1305 43697363 6F312030 1E060355 04031317 43697363
6F204C69 63656E73 696E6720 526F6F74 20434130 1E170D31 33303533 30313934
3834375A 170D3338 30353330 31393438 34375A30 32310E30 0C060355 040A1305
43697363 6F312030 1E060355 04031317 43697363 6F204C69 63656E73 696E6720
526F6F74 20434130 82012230 0D06092A 864886F7 0D010101 05000382 010F0030
82010A02 82010100 A6BCBD96 131E05F7 145EA72C 2CD686E6 17222EA1 F1EFF64D
CBB4C798 212AA147 C655D8D7 9471380D 8711441E 1AAF071A 9CAE6388 8A38E520
1C394D78 462EF239 C659F715 B98C0A59 5BBB5CBD 0CFEBEA3 700A8BF7 D8F256EE
4AA4E80D DB6FD1C9 60B1FD18 FFC69C96 6FA68957 A2617DE7 104FDC5F EA2956AC
7390A3EB 2B5436AD C847A2C5 DAB553EB 69A9A535 58E9F3E3 C0BD23CF 58BD7188
68E69491 20F320E7 948E71D7 AE3BCC84 F10684C7 4BC8E00F 539BA42B 42C68BB7
C7479096 B4CB2D62 EA2F505D C7B062A4 6811D95B E8250FC4 5D5D5FB8 8F27D191
C55F0D76 61F9A4CD 3D992327 A8BB03BD 4E6D7069 7CBADF8B DF5F4368 95135E44
DFC7C6CF 04DD7FD1 02030100 01A34230 40300E06 03551D0F 0101FF04 04030201
--More-- 06300F06 03551D13 0101FF04 05300301 01FF301D 0603551D 0E041604 1449DC85
4B3D31E5 1B3E6A17 606AF333 3D3B4C73 E8300D06 092A8648 86F70D01 010B0500
03820101 00507F24 D3932A66 86025D9F E838AE5C 6D4DF6B0 49631C78 240DA905
604EDCDE FF4FED2B 77FC460E CD636FDB DD44681E 3A5673AB 9093D3B1 6C9E3D8B
D98987BF E40CBD9E 1AECA0C2 2189BB5C 8FA85686 CD98B646 5575B146 8DFC66A8
467A3DF4 4D565700 6ADF0F0D CF835015 3C04FF7C 21E878AC 11BA9CD2 55A9232C
7CA7B7E6 C1AF74F6 152E99B7 B1FCF9BB E973DE7F 5BDDEB86 C71E3B49 1765308B
5FB0DA06 B92AFE7F 494E8A9E 07B85737 F3A58BE1 1A48A229 C37C1E69 39F08678
80DDCD16 D6BACECA EEBC7CF9 8428787B 35202CDC 60E4616A B623CDBD 230E3AFB
418616A9 4093E049 4D10AB75 27E86F73 932E35B5 8862FDAE 0275156F 719BB2F0
D697DF7F 28
quit
!
!
!
!
voice service voip
ip address trusted list
ipv4 192.168.2.1
ipv4 192.168.2.2
ipv4 192.168.1.0 255.255.255.0
media disable-detailed-stats
allow-connections sip to sip
trace
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
registrar server
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
!
!
!
!
voice class tenant 100
registrar ipv4:192.168.2.1 expires 3600
sip-server ipv4:192.168.2.1
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
!
!
!
voice register global
mode cme
source-address 192.168.1.5 port 5060
max-dn 40
max-pool 50
tftp-path flash:
file text
create profile sync 0183265509184109
auto-register
!
!
voice register dn 1
number 100
!
voice register dn 2
number 101
!
!
voice register dn 11
number XXX3789XX

!
voice register pool 11
busy-trigger-per-button 2
id mac EC01.D50B.F0AF
type 7841
number 1 dn 11
no digit collect kpml

!
!
voice translation-rule 1
rule 1 /^.*/ /XXX3789XX/
!
!
voice translation-profile PSTN-Out
translate calling 1
!
redundancy
mode none
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0/0
ip address 192.168.1.5 255.255.255.0
negotiation auto
!
interface GigabitEthernet0/0/1
ip address 192.168.2.2 255.255.255.0
negotiation auto
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
shutdown
--More-- negotiation auto
!
ip default-gateway 192.168.1.1
ip http server
ip http authentication local
ip http secure-server
ip forward-protocol nd
ip tftp source-interface GigabitEthernet0/0/0
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
!
!
!
!
control-plane
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 1 voip
destination-pattern .T
session target sip-server
voice-class sip tenant 100
no voice-class sip pass-thru content sdp
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
codec g711alaw
!
dial-peer voice 2 voip
session protocol sipv2
incoming called-number .
codec g711alaw
no vad
!
!
!
line con 0
Cisco-4321#
Cisco-4321#
Cisco-4321#
Cisco-4321#

Roger Kallberg · ‎09-15-2022

You are getting this for your call attempts.

*Sep 15 13:37:02.975: //325/50F66C608273/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x7FA893509D38, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=XXX2165XXX,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=XXX3789XX(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=40001, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Sep 15 13:37:02.977: //325/50F66C608273/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.1;branch=z9hG4bK-1F13-142
From: <sip:XXX2165XXX@192.168.2.1;user=phone>;tag=408D
To: <sip:XXX3789XX@192.168.2.2;user=phone>;tag=765279-9E0
Date: Thu, 15 Sep 2022 13:37:02 GMT
Call-ID: OA6AB51491XXX2165XXX90DC72C6
CSeq: 324 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.1.5 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-17.3.4a
Reason: Q.850;cause=47
Session-ID: 00000000000000000000000000000000;remote=5668dfd7e5cf5900840dac98116258d5
Content-Length: 0

In your configuration there are a number of things that I can see are incorrect, I’ll get back to you on this with details once I’m in front of a computer, it’s simply not practical to do so on a mobile device.

Roger Kallberg · ‎09-15-2022

Try with adding these changes to your configuration. Not all things are related to the SIP trunk, some are general clean-up.

voice service voip
ip address trusted list
 no ipv4 192.168.2.2 ! Not needed as you don't need allow traffic from the router it self.
 no ipv4 192.168.1.0 255.255.255.0 ! Not needed as you don't need allow traffic from phones to the router, these are explicitly allowed by the CME dial peers.
mode border-element ! Turns on CUBE functionallity.
!
voice register pool 11
 voice-class codec 1 ! Set use of the codec list for the CME phone, without this it will use g729 that is the default if there is no configuration.
 no vad ! Turns off VAD.
 dtmf-relay sip-notify rtp-nte ! Defines what dtmf-relay function to use.
!
voice class codec 1
 no codec preference 4 g729br8 ! This codec involves VAD as it has a "b" in the name and that is something you should stay away from in general.
!
no ip default-gateway 192.168.1.1 ! Not needed, this is for a switched enviroment and you have a routed.
!
voice class uri PSTN sip ! Use information in VIA header to match the inbound dial peer.
 host ipv4:<ITSP IP as seen in VIA header in the invite for inbound calls from PSTN>
!
no voice translation-rule 1 ! Not needed as it is not in use in your current configuration.
!
voice translation-rule 40 ! Translate called number to PSTN to drop the breakout code.
 rule 1 /^0\(.*\)/ /\1/ ! Or /^9\(.*\)/ /\1/ if you use that as the PSTN breakout.
!
voice translation-profile PSTN-Out
 no translate calling 1 ! Not needed as it was not in use.
 translate called 40 ! Translate called number to PSTN to drop the breakout code.
!
dial-peer voice 1 voip
 description Outbound calls to PSTN
 translation-profile outgoing PSTN-Out ! Calls translate profile to drop the breakout code.
 session protocol sipv2 ! You had not defined SIP as the protocol on this dial peer and then it uses H.323 as that is the default
 destination-pattern .T ! Set this to something that is more specific and only matches outbound calls to PSTN. For example 0T or 9T if you use that as a breakout code for calls to PSTN.
 no session target sip-server ! Use session target ipv4 instead.
 session target ipv4:192.168.2.1 ! Set the session target to your ITSP router. I'm not sure if this IP is what you should use, you'd have to check that information with the service provider.
 no voice-class sip tenant 100 ! Not needed as you don't use autentication or registration.
 no codec g711alaw ! Remove use of just one codec.
 voice-class codec 1 ! Use codec list.
 no vad ! Turns off VAD
 default voice-class sip audio forced ! Removes video negotionation from the SDP
 dtmf-relay rtp-nte ! Defines what dtmf-relay function to use.
!
dial-peer voice 2 voip
 description Inbound calls from PSTN
 no incoming called-number . ! Not a good match for inbound dial peer.
 incoming uri via PSTN ! Use information in VIA header to match the inbound dial peer.
 no codec g711alaw ! Remove use of just one codec.
 voice-class codec 1 ! Use codec list.
 voice-class sip bind control source-interface GigabitEthernet0/0/1 ! Add bind to inbound dial peer.
 voice-class sip bind media source-interface GigabitEthernet0/0/1 ! Add bind to inbound dial peer.
 dtmf-relay rtp-nte ! Defines what dtmf-relay function to use.
!
no voice class tenant 100 ! Not needed as you don't use autentication or registration.
!
sip-ua
 registrar ipv4:192.168.1.5 expires 3600 ! Used by CME SIP phones.

Rajan R · ‎09-15-2022

Any luck on this ?