SIP Trunk won't register. Won't allow calls out or in. Going crazy!!! Could you help, pls?

[LuisReis08240]

Hello Community,

I've got a IP7970, registered on a CME (on a c892). I've got an SIP number from my ISProvider, along with the sip-server, registrar and credentials. All works on a soft-phone, but for the life of me, I'm not able to have it register on the SIP server, nor can I have it call outside, by adding the sip server (and outbound-proxy) on the dial-peer...

My setup is: IP7970-->CME(c892)-->catalyst2960G-->ISPModem.

I have no VLANs set up, and am using the same ip range for all the devices on my network (basic: 192.168.1.0/24)

I'm going crazy, as I've spent days and days trying possible configurations and reading our community posts up and down...

Here goes my current config:

Current configuration : 4909 bytes
!
! Last configuration change at 16:52:58 central Sun Nov 15 2020
version 15.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname siprouter
!
boot-start-marker
boot-end-marker
!
!
enable secret
enable password
!
no aaa new-model
clock timezone central 1 0
clock summer-time central recurring
!
!
ip cef
!
!
!
!

!
!
!
!
ip domain name casadelrei.com
ip name-server 1.1.1.1
ip name-server 192.168.1.14
no ipv6 cef
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
signaling forward none
fax sip
bind control source-interface GigabitEthernet0
bind media source-interface GigabitEthernet0
rel1xx disable
outbound-proxy dns:proxy.sip.sapo.pt:5070
sip
bind control source-interface GigabitEthernet0
bind media source-interface GigabitEthernet0
rel1xx disable

outbound-proxy dns:proxy.sip.sapo.pt:5070
!
voice class codec 1
codec preference 1 g711ulaw
!
!
voice register global
mode cme
source-address 192.168.1.26 port 5060
max-dn 5
max-pool 3
load 7970 SCCP70.9-2-1S.loads
tftp-path flash:
create profile sync 0009111433217796
!
voice register pool 1
id mac 001B.5452.C01D
type 7970
dtmf-relay sip-notify
codec g711ulaw
!
!
voice translation-rule 1
rule 1 // /00351/
!
!
voice translation-profile ADD00351*
translate called 1
!
!
license udi pid CISCO892-K9
!
!
file privilege 0
username
!
redundancy
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface BRI0
no ip address
encapsulation hdlc
shutdown
isdn termination multidrop
!
interface FastEthernet0
no ip address
shutdown
!
interface FastEthernet1
no ip address

shutdown
!
interface FastEthernet2
no ip address
shutdown
!
interface FastEthernet3
no ip address
shutdown
!
interface FastEthernet4
no ip address
shutdown
!
interface FastEthernet5
no ip address
shutdown
!
interface FastEthernet6
no ip address
shutdown
!
interface FastEthernet7
no ip address
shutdown
!
interface FastEthernet8
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0
ip address 192.168.1.26 255.255.255.0
duplex auto
speed auto
!
interface Vlan1
no ip address
shutdown
!
ip default-gateway 192.168.1.254
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
!
!
!
tftp-server term70.default.loads
tftp-server term71.default.loads
tftp-server apps70.8-5-2TH1-9.sbn
tftp-server cnu70.8-5-2TH1-9.sbn
tftp-server cvm70sccp.8-5-2TH1-9.sbn
tftp-server jar70sccp.8-5-2TH1-9.sbn
tftp-server SCCP70.8-5-2S.loads
tftp-server apps70.9-2-1TH1-13.sbn
tftp-server cnu70.9-2-1TH1-13.sbn
tftp-server cvm70sccp.9-2-1TH1-13.sbn
tftp-server dsp70.9-2-1TH1-13.sbn

tftp-server jar70sccp.9-2-1TH1-13.sbn
tftp-server SCCP70.9-2-1S.loads
!
control-plane
!
!
!
!
mgcp profile default
!
!
!
dial-peer voice 2 voip
description ** Outgoing Call to SIP Trunk**
destination-pattern .T
session protocol sipv2
session target sip-server
session transport udp
no voice-class sip outbound-proxy
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
credentials number +351302004889 username +351302004889@sip.sapo.pt password XXXXXXX realm sip.sapo.pt
authentication username +351302004889 password XXXXXXXX realm sip.sapo.pt
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.sapo.pt expires 3600
sip-server dns:sip.sapo.pt
connection-reuse
host-registrar
!
!
telephony-service
max-ephones 3
max-dn 4 no-reg both
ip source-address 192.168.1.26 port 2000
calling-number initiator
service phone webAccess 0
cnf-file location flash:
load 7970 SCCP70.8-5-2S.loads
time-format 24
date-format dd-mm-yy
voicemail 00351200
max-conferences 0 gain -6
web admin system name
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1

!
telephony-service
max-ephones 3
max-dn 4 no-reg both
ip source-address 192.168.1.26 port 2000
calling-number initiator
service phone webAccess 0
system message Welcome Home
cnf-file location flash:
load 7970 SCCP70.8-5-2S.loads
time-format 24
date-format dd-mm-yy
voicemail 00351200
max-conferences 0 gain -6
web admin system name g0d secret 5 $1$LrSn$lzixDjSNzcZMZxDkBVww2/
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1
!
!
ephone-dn 1
number 800 no-reg both
description Office Phone
name Escritorio
hold-alert 30 originator
!
!
ephone-dn 2 dual-line
number +351302004489 no-reg both
label SIP Number
description SIP Number
name SIP
hold-alert 30 originator
!
!
ephone 1
device-security-mode none
mac-address 001B.5452.C01D
after-hours exempt
type 7970
button 1:1 2:2
!
!
!
banner motd ^CAdmin Access only!^C
!
line con 0
line aux 0
line vty 0 4
login
transport input all
!
ntp update-calendar
ntp server 1.pt.pool.ntp.org prefer
!
end

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

Here goes my debug ccsip all and then I've made a call that gave nothing but silence, followed by a busy signal and a "Ring out" message on the phone:

siprouter#term mon
siprouter#debug ccsip all
This may severely impact system performance. Continue? [confirm]
All SIP Call tracing is enabled
siprouter#
*Nov 15 16:15:31.756: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x8C774DD8) with key=[219] to table
*Nov 15 16:15:31.756: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
*Nov 15 16:15:31.756: //-1/000000000000/SIP/Info/ccsip_iwf_init:
*Nov 15 16:15:31.756: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
*Nov 15 16:15:31.756: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization: Entry...
*Nov 15 16:15:31.756: //-1/000000000000/SIP/Info/ccsip_ipip_media_forking_init: MF: Queue is initialised..
*Nov 15 16:15:31.756: //232/000000000000/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_call_setup_request: Set Protocol information
*Nov 15 16:15:31.756: //232/000000000000/SIP/Error/ccsip_ipip_media_forking_read_from_TDContainer:
MF: Unable to read data from TD Container..
*Nov 15 16:15:31.756: //232/000000000000/SIP/Error/ccsip_ipip_media_forking_forked_leg_config:
MF: TD container cannot be read/container is NULL. Setting of forked call leg failed..
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_handle_peer_event:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_map_ccapi_event_to_iwf_event: Event Category: 1, Event Id: 183
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_process_event:
*Nov 15 16:15:31.756: //232/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:cur_container:sip_iwf_default_early_dialog_container, cur_state:S_SIP_IWF_SDP_IDLE, event:E_SIP_IWF_EV_SET_MODE
*Nov 15 16:15:31.756: //232/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_sip_md_snr:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_sip_ed_snr:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_sip_md:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_sip_ed:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_h32x_in_set_mode:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_h323_in_set_mode:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_sccp_in_set_mode:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/is_mode_sip_sccp_in_set_mode:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/sip_iwf_def_set_mode_hdlr: Setting SPI mode to SIP-TDM
*Nov 15 16:15:31.756: //232/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_sccp_early_dialog_container
*Nov 15 16:15:31.756: //232/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:next_state:CNFSM_NO_STATE_CHANGE
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_handle_peer_event: Return value : SIP_IWF_SUCCESS
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_handle_peer_event:
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_map_ccapi_event_to_iwf_event: Event Category: 3, Event Id: 5
*Nov 15 16:15:31.756: //232/000000000000/SIP/Info/ccsip_iwf_process_event:
*Nov 15 16:15:31.756: //232/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:cur_container:sip_iwf_sccp_early_dialog_container, cur_state:S_SIP_IWF_SDP_IDLE, event:E_SIP_IWF_EV_SET_FLOW_MODE
*Nov 15 16:15:31.760: //232/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/is_fa2ft_md_flow_mode_transition:
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/is_fa2ft_flow_mode_transition:
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/ccsip_get_flow_mode_frm_set_flow_mode_ev:
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/is_fa2ft_flow_mode_transition:
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/ccsip_get_flow_mode_frm_set_flow_mode_ev:
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/ccsip_iwf_process_event: IWF - cnfsm ret 2
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/ccsip_iwf_handle_peer_event: Return value : SIP_IWF_SUCCESS
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/ccsip_call_setup_request: Before processing SETUP REQccb->pld.flags_ipip = 200
*Nov 15 16:15:31.760: //232/000000000000/SIP/Info/ccsip_call_setup_request: After processing SETUP REQccb->pld.flags_ipip = 200
*Nov 15 16:15:31.760: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : sip.sapo.pt target_port : 5060

*Nov 15 16:15:31.760: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsip_call_setup_request: Incrementing call counter to [1] in dial-peer [2]
*Nov 15 16:15:31.760: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 2
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id E8 to table
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: preferred_codec set[0] type :No Codec bytes: 0
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPI_ipip_store_config_info: Setting mid_call_config_info = 0x0 for callid = 232
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: Media forking disabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=231
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: Incoming: No defer BYE for last
call stats
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: Media forking disabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_anchor_leg_config: MF: en_p->encap_s.voIP.voipPeerCfgMediaClass = 0
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_anchor_leg_config: MF: Dial-peer has no media class recorder.
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIMFChangeState: MF: Prev state = 0 & New state = -1
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_anchor_leg_reset: MF: Anchor leg config reset done...
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsipInitDSCPPolicyInfo: No DSCP Profile configured, No RPH 2 DSCP Mapping and DSCP policing
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIGetCallConfig: Initilise the DSCP policy
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/preprocessSetup:
This is a not a SIGO Call -, could be DM call
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsip_iwf_process_event:
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/State/ccsip_cnfsm_debugs: IWF:cur_container:sip_iwf_sccp_early_dialog_container, cur_state:S_SIP_IWF_SDP_IDLE, event:E_SIP_IWF_EV_INIT_CALL_SETUP
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/ccsip_iwf_process_event: IWF - cnfsm ret 2
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/preprocessSetup: SIP-TDM or TCL/VXML app case
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sip_gw_pre_setup_update_stream_media_direction: peer_callID = 231
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sip_gw_pre_setup_update_stream_media_direction: peer_channels/stream is NULL
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sip_gw_pre_setup_add_sdp_container: DNS/ENUM resolution required; Deferred Creating SDP
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIBwCacUpdateAccountedBw: bwcac update accounted BW Option 0 flow mode flow-through
audio bw 0 bps video bw 0 bps fax bw 0 bps total bw 0 bps accounted bw 0 bps
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIBwCacUpdateInterfaceBw: bwcac acquiring interface GigabitEthernet0 bw 0
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIBwCacUpdateAccountedBw: bwcac update accounted bw (initial offer) accounted bw set to 0
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIBwCacIsDialPeerBwAvailable: bwcac NOP dial-peer bw available tag 2
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIBwCacIsInterfaceBwAvailable: bwcac interface bw threshold not configured
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIBwCacVerifyBwThreshold: bwcac verify bw threshold, bw available allow call total bw 0 bps
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIValidateGtd: Signal Forward disabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIValidateTunnelData: RawMsg/QSIG Tunneling Not Enabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIAddMLPPServicesInfo: No MLP Info available on incoming leg
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIPreprocessUriFormat: Url cfg for 1: 2,phone-ctxt=FALSE
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIAddCiscoGcid: Gcid value not set - not adding header.
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPIAddPrivacyandIdentityInfo: ccb->local_host_name,ccb->src_addr_str is NULL
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Info/sipSPICompareHistoryInfoWithMatchedDialpeer: call-route history-info CLI not enabled
*Nov 15 16:15:31.760: //232/9662CC508052/SIP/Error/sipSPI_ipip_set_history_info_header:
ccb->src_addr_str is NULL
*Nov 15 16:15:31.764: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE
*Nov 15 16:15:31.764: //232/9662CC508052/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_SENT_DNS)
*Nov 15 16:15:31.764: //232/9662CC508052/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_IDLE, SUBSTATE_SENT_DNS) to (STATE_IDLE, SUBSTATE_SENT_DNS)
*Nov 15 16:15:31.764: //-1/xxxxxxxxxxxx/SIP/Info/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 192.168.1.26 for SIP
*Nov 15 16:15:31.764: //232/9662CC508052/SIP/Info/resolve_sig_ip_address_to_bind: signaling bind address : 192.168.1.26
*Nov 15 16:15:31.764: //232/9662CC508052/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
*Nov 15 16:15:31.764: //232/9662CC508052/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x8C774DD8 key=9E4AB87A-269411EB-8057F911-9F6D9FE1@192.168.1.26
*Nov 15 16:15:31.764: //232/9662CC508052/SIP/Info/sipSPIUsetBillingProfile: sipCallId for billing records = 9E4AB87A-269411EB-8057F911-9F6D9FE1@192.168.1.26
*Nov 15 16:15:31.764: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.sip.sapo.pt and type:1
*Nov 15 16:15:49.764: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: DNS query for sip.sapo.pt and type:1
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: TYPE A query successful for sip.sapo.pt
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: IP Address of sip.sapo.pt is:

*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_aaaa_query: 213.13.145.45

*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 43
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_IDLE, SUBSTATE_SENT_DNS) to (STATE_IDLE, SUBSTATE_NONE)
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICacheHostToCCB: sipSPICacheHostToCCB dnsResponse.num_hosts = 1
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICacheHostToCCB: IP Address No. 1, IP address 213.13.145.45
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 192.168.1.26 for SIP
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/resolve_sig_ip_address_to_bind: signaling bind address : 192.168.1.26
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 192.168.1.26 for SIP
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/resolve_sig_ip_address_to_bind: signaling bind address : 192.168.1.26
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPICompareHistoryInfoWithMatchedDialpeer: call-route history-info CLI not enabled
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPI_ipip_set_history_info_header: No HI header recvd from container
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIAddPrivacyandIdentityInfo: Removing "id" value from Privacy
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 192.168.1.26 for SIP
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.26
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_add_forking_stream: MF: Not a forked SPI leg..
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_update_forking_stream: MF: Not a forked SPI leg..
SIP: (232) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 192.168.1.26 for SIP
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/resolve_sig_ip_address_to_bind: signaling bind address : 192.168.1.26
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16402 for stream 1
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIDoBearerCapToCodecMapping: Bearer capability to Codec Mapping: DISABLED

*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacCalcMaxAudioBandwidth: calculating max bw from preffered codecs (local offer)
SIP: (232) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacCalcMaxAudioBandwidth: max bw (excluding pak overhead) from preffered codecs: codec g711ulaw bw 64000 index 0
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacCalcMaxAudioBandwidth: audio caps channel idx not found !!!!
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacCalcMaxAudioBandwidth: max bw (including pak overhead) from preffered codecs: codec g711ulaw bw 80000
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIOutgoingCallSDP: Creating recv-only stream for outbound call
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacUpdateAccountedBw: bwcac update accounted BW Option 5 flow mode flow-through
audio bw 80000 bps video bw 0 bps fax bw 0 bps total bw 80000 bps accounted bw 0 bps
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacUpdateInterfaceBw: NOP (no interface change)
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIBwCacUpdateAccountedBw: bwcac update accounted bw (no interface change)
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 213.13.145.45,Port 5060, Transport 1, SentBy Port 5060
*Nov 15 16:16:07.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone central to SIP default timezone = GMT
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:8E24FC94
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/Session-Timer/sipSTSLSRReqSend: Session timer is not required
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/Session-Timer/sipSTSLMain:
SE: 0;refresher:none peer refresher:none, flags:2000, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 0 event
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Nov 15 16:16:07.836: //232/9662CC508052/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_offer_ans_handle_sent_sdp:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_offer_ans_process_event:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/State/ccsip_cnfsm_debugs: OA:cur_container:ccsip_offer_ans_main_container, cur_state:S_SIP_EARLY_DIALOG_IDLE, event:E_SIP_INVITE_SDP_SENT
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_offer_ans_is_invite_offer_valid: - 1
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_offer_ans_common_offer_sent_hdlr:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_iwf_handle_network_event:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_iwf_process_event:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/State/ccsip_cnfsm_debugs: IWF:cur_container:sip_iwf_sccp_early_dialog_container, cur_state:S_SIP_IWF_SDP_IDLE, event:E_SIP_IWF_EV_SENT_SDP
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/is_sent_sccp_do_video_inactive:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/is_mode_sip_sccp_do_video:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/is_mode_sip_sccp_do_video:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sip_iwf_def_ed_sent_sdp_offer_hdlr:
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/State/ccsip_cnfsm_debugs: IWF:next_state:S_SIP_IWF_SDP_SENT_AWAIT_SDP
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/State/ccsip_cnfsm_debugs: OA:next_state:S_SIP_EARLY_DIALOG_OFFER_SENT
*Nov 15 16:16:07.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIgetRegistrarHost: registrar host retrieved : sip.sapo.pt
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_get_forked_recording_data: MF: Not an Forked leg..
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPICreateRecParticipantHeaders: X-Cisco-Recording-Participant header not added.
SIP: (232) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp dont have anat
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPISendInvite: Associated container=0x8E24FC94 to Invite
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Transport/sipSPITransportSendMessage: msg=0x8E23B060, addr=213.13.145.45, port=5060, sentBy_port=0, local_addr=192.168.1.26, is_req=1, transport=1, switch=0, callBack=0x804CB8B8
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Nov 15 16:16:07.840: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportLogicSendMsg: connection-reuse configured, listen conn-id : 2
*Nov 15 16:16:07.840: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8E23B060, addr=213.13.145.45, port=5060, local_addr=192.168.1.26, connId=2 for UDP
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sentInviteRequest: Sent Invite in state STATE_IDLE
*Nov 15 16:16:07.840: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteRequest: Transaction active. Facilities will be queued.
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_SENT_INVITE, SUBSTATE_NONE)
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIAddStream:
set stream_callid from ccb->ccCallID:0xE8, media_type:0
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 232) to the VOIP RTP library
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.26
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIUpdateRtcpSession:
ccb->flags != LOOPBACK

*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 192.168.1.26, lport = 16402, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 232, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPICreateRtpSession: sess: 8DDE4528 do_rtcp:0
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIUpdateRtcpSession: Not able to Associate DSCP Profile with GCCB dscp_policy = 0x0, IS_SIPSPI_MODE_IN_SIP_SIP = 0 dscpPolicySeviceBlock = 0x0 , stream->qos_info = 0x0
*Nov 15 16:16:07.840: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Nov 15 16:16:07.840: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=232
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIUpdateRtcpSession:
DTMF inb/oob disabled
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/State/sipSPIChangeStreamState: Stream (callid = 232) State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Info/sipSPIUpdateCallEntry:
Call 232 set InfoType to SPEECH
*Nov 15 16:16:07.840: //232/9662CC508052/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:00351924395857@sip.sapo.pt:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bKD9206A
From: "Escritorio" <sip:800@sip.sapo.pt>;tag=30A1450-13CE
To: <sip:00351924395857@sip.sapo.pt>
Date: Sun, 15 Nov 2020 16:16:07 GMT
Call-ID: 9E4AB87A-269411EB-8057F911-9F6D9FE1@192.168.1.26
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2523057232-0647238123-2152921361-2674761697
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1605456967
Contact: <sip:800@192.168.1.26:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 6573 3653 IN IP4 192.168.1.26
s=SIP Call
c=IN IP4 192.168.1.26
t=0 0
m=audio 16402 RTP/AVP 0 101
c=IN IP4 192.168.1.26
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Nov 15 16:16:08.340: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone central to SIP default timezone = GMT
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_get_forked_recording_data: MF: Not an Forked leg..
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Info/sipSPICreateRecParticipantHeaders: X-Cisco-Recording-Participant header not added.
SIP: (232) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp dont have anat
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Info/sipSPISendInvite: Associated container=0x8E24FC94 to Invite
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Transport/sipSPITransportSendMessage: msg=0x8E23B060, addr=213.13.145.45, port=5060, sentBy_port=0, local_addr=192.168.1.26, is_req=1, transport=1, switch=0, callBack=0x0
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Nov 15 16:16:08.340: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportLogicSendMsg: connection-reuse configured, listen conn-id : 2
*Nov 15 16:16:08.340: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8E23B060, addr=213.13.145.45, port=5060, local_addr=192.168.1.26, connId=2 for UDP
*Nov 15 16:16:08.340: //232/9662CC508052/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:00351924395857@sip.sapo.pt:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bKD9206A
From: "Escritorio" <sip:800@sip.sapo.pt>;tag=30A1450-13CE
To: <sip:00351924395857@sip.sapo.pt>
Date: Sun, 15 Nov 2020 16:16:08 GMT
Call-ID: 9E4AB87A-269411EB-8057F911-9F6D9FE1@192.168.1.26
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2523057232-0647238123-2152921361-2674761697
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1605456968
Contact: <sip:800@192.168.1.26:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 6573 3653 IN IP4 192.168.1.26
s=SIP Call
c=IN IP4 192.168.1.26
t=0 0
m=audio 16402 RTP/AVP 0 101
c=IN IP4 192.168.1.26
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Nov 15 16:16:09.340: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone central to SIP default timezone = GMT
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_get_forked_recording_data: MF: Not an Forked leg..
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Info/sipSPICreateRecParticipantHeaders: X-Cisco-Recording-Participant header not added.
SIP: (232) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp dont have anat
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Info/sipSPISendInvite: Associated container=0x8E24FC94 to Invite
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Transport/sipSPITransportSendMessage: msg=0x8E23B060, addr=213.13.145.45, port=5060, sentBy_port=0, local_addr=192.168.1.26, is_req=1, transport=1, switch=0, callBack=0x0
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Nov 15 16:16:09.340: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportLogicSendMsg: connection-reuse configured, listen conn-id : 2
*Nov 15 16:16:09.340: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8E23B060, addr=213.13.145.45, port=5060, local_addr=192.168.1.26, connId=2 for UDP
*Nov 15 16:16:09.340: //232/9662CC508052/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:00351924395857@sip.sapo.pt:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bKD9206A
From: "Escritorio" <sip:800@sip.sapo.pt>;tag=30A1450-13CE
To: <sip:00351924395857@sip.sapo.pt>
Date: Sun, 15 Nov 2020 16:16:09 GMT
Call-ID: 9E4AB87A-269411EB-8057F911-9F6D9FE1@192.168.1.26
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2523057232-0647238123-2152921361-2674761697
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1605456969
Contact: <sip:800@192.168.1.26:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 6573 3653 IN IP4 192.168.1.26
s=SIP Call
c=IN IP4 192.168.1.26
t=0 0
m=audio 16402 RTP/AVP 0 101
c=IN IP4 192.168.1.26
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Error/act_sentinvite_wait_100:
Out of retries
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIInitInvTryOtherIP: no more destinations in list
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_set_cc_cause_for_spi_err: Categorized cause:102, category:129
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[232], src[6]
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(102) for outgoing call
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG:
No Inbound Container Created !!!
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931:
No Inbound Container Created !!!
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_ipip_media_service_get_event_data: Event id = 28
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPI_ipip_antiTrombone: Entered Antitrombone service
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_post_event: MF: Not a Anchor SIP leg..
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_call_statistics: Requesting stats for callid=232
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_call_statistics: Stats request failed for callid=232, dstCallID=-1, rc=-7
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_ipip_media_service_get_event_data: Event id = 28
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPI_ipip_antiTrombone: Entered Antitrombone service
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_ipip_media_forking_post_event: MF: Not a Anchor SIP leg..
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 8
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SPI_EVENT
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/Session-Timer/sipSTSLMain:
SE: 0;refresher:none peer refresher:none, flags:2000, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/act_disconnecting_disconnect: Disconnect now.. no defer BYE..
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIDeferCallClose: Not split dataplane, bail
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIWaitForStatsBforeCallClose: Not split dataplane, bail
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPICallCloseAfterFinalStat:
sipSPICallCloseAfterFinalStat:
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:5099570 ConnTime 0
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8C54242C
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/State/sipSPIChangeState: 0x8C774DD8 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x8C774DD8
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 800
Called Number : 00351924395857
Source IP Address (Sig 192.168.1.26
Destn SIP Req Addr:Port : 213.13.145.45:5060
Destn SIP Resp Addr:Port : 213.13.145.45:5060
Destination Name : sip.sapo.pt

*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.1.26
Source IP Port (Media): 16402
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 102
Disconnect Cause (SIP) : 408

*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id E8
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[219] removed.
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x8C774DD8 key=9E4AB87A-269411EB-8057F911-9F6D9FE1@192.168.1.26
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerUnregisterCtxtInConnection: gConnTab=0x8BE083A4, addr=213.13.145.45, port=5060, local_addr=192.168.1.26, unregistering context=0x8C774DD8
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Error/sipConnectionManagerUnregisterCtxtInConnection:
Could not find conn holder for addr=213.13.145.45
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Transport/sipSPITransportContextCleanup: Could not purge context gcb=0x8C774DD8 from the connection; gcb might be locked
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_offer_ans_delete:
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/ccsip_iwf_delete:
*Nov 15 16:16:11.340: //232/9662CC508052/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 0x8C774DD8
*Nov 15 16:16:11.340: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[219]

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

Thank you so much for helping me with this!!!

Kind Regards,

Luís

[LuisReis08240]

I've tested with the other "line", that has the ISP provided number configured as the DN number.

Attached.

Thank you!

[pajacek]

Hi,

First you should register IP Phone to CME.. Than you should register trunk to ITSP, and now I don't think your trunk is registered to ITSP system. I don't know about informations you recieved from ITSP needed to register trunk but from my perspective:

1. Sip-Ua config should be like that:

sip-ua
credentials number +351302004889 username +351302004889 password XXXXXXX realm sip.sapo.pt
authentication username +351302004889 password XXXXXXXX realm sip.sapo.pt

no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.sapo.pt expires 3600
sip-server dns:sip.sapo.pt
connection-reuse
host-registrar

or even like that: (I'm not sure there should be +.. Please check credentials you recieved from ITSP)

sip-ua
credentials number 351302004889 username 351302004889 password XXXXXXX realm sip.sapo.pt
authentication username 351302004889 password XXXXXXXX realm sip.sapo.pt

no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.sapo.pt expires 3600
sip-server dns:sip.sapo.pt
connection-reuse
host-registrar

2. Are you sure SIP port should be 5070 and not standard 5060?

sip

 outbound-proxy dns:proxy.sip.sapo.pt:5060

3. Outgoing dial-peer:

dial-peer voice 2 voip
description ** Outgoing Call to SIP Trunk**
destination-pattern .T
session protocol sipv2
session target sip-server
session transport udp
voice-class sip outbound-proxy dns:proxy.sip.sapo.pt

voice-class sip bind control source-interface GigabitEthernet0
voice-class sip bind media source-interface GigabitEthernet0
dtmf-relay rtp-nte
codec g711ulaw
no vad
!

Some other problems... Have your 7970 Phone some IP address?? Is this phone registered to CME? It's a SIP or SCCP phone?? Where is your phone connected to, because as I see all FastEthernet ports are in shutdown state...

I think there is more issues in your config... what I did before could only start to help with SIP trunk.

Regards,

Jack

[LuisReis08240]

Hi Jack,

Thank you very much for your reply and terribly sorry for taking so long to get back to you. These are weird times we are living and didn't have the time to get back to this problem. It's still driving me crazy, btw

I've tried changing the "+" sign and checked with my ISP for the out-bound proxy port. 5070 is the correct one for my ISP.

The "+" sign should work, but it does not. I've tried with some soft-phone software and it works. Just not on my Cisco Router config

Answering your questions:

- Yes, the 7970 has an IP address

- It is registered on the CME

- Its SCCP.

- It's connected to a Cat.2960G, as is the router, and a BroadBand Router (Def. GW).

I'm really running out of ideas... 

The router won't register to the sip trunk:

Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
+351302004889 -1 0 no

Thanks again for your help and for your time.

Kind Regards,

Luís

[b.winter · Accepted solution]

Hi,

As you are using DNS-names as SIP-registrar and outbound-proxy, can the router resolve those entries? (do you see them in "show hosts"?)

Have you checked the logs for the SIP registration messages? (with "debug ccsip all" or "debug ccsip non-call")

Since the SIP trunk is not registered, I would start troubleshooting this problem.

Before that, you won't be able to make calls.

Also, I would recommend using tenant configuration. So, configure SIP trunk details in a tenant and not into the "sip-ua" section, then bind the tenant to the dial-peers.

Best regards,

Björn