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outbound call fail using SIP Trunk on CME

moustafa.idc
Level 1
Level 1

Hi all,

 

I have Cisco CME with some SCCP & SIP phones. I have also sip provider to route local and international calls to.

When I configured the router for sip phones, outbound call stopped working. I'm still able to make inbound calls through sip trunk.

Attached is the config and traces.

 

appreciate if somebody can look at it and see if I missed something.

 

Thanks,

 

1 Accepted Solution

Accepted Solutions

Try to move the ITSP registrar bit to a tenant configuration. This would be an example of how that could look like, you'd need to alter it to fit your specifics.

voice class tenant 2000
  registrar dns:<service provider DNS name> expires 3600
  credentials username <user name for connection> password 7 <password for connection> realm <service provider realm name>
  authentication username <user name for connection> password 7 <password for connection>
  no remote-party-id
  timers dns registrar-cache 95
  sip-server dns:<service provider DNS name>
  connection-reuse
  audio forced
  bind control source-interface GigabitEthernet0/0/1
  bind media source-interface GigabitEthernet0/0/1
  no pass-thru content custom-sdp
  sip-profiles 10
  outbound-proxy dns:<service provider DNS name for proxy> reuse
  early-offer forced


dial-peer voice 100 voip
 description Inbound calls from PSTN
 translation-profile incoming PSTN-IN
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 2000  
 voice-class sip tenant 2000
 dtmf-relay rtp-nte
 no vad   
!
dial-peer voice 110 voip
 description Outbound calls to PSTN
 translation-profile outgoing PSTN-OUT
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 2000  
 voice-class sip tenant 2000
 voice-class sip options-keepalive profile 2000
 dtmf-relay rtp-nte
 no vad

The reason I think this is affecting you is that SIP phones need to have a registrar under sip-ua, so it interference with what you have for the ITSP.

 

This is from a SBC where we run a SIP connection that uses registration and we're having SRST for SIP phones on it as well. As SRST and CME is pretty much the same thing when it comes to this part it should fit.

voice service voip
ip address trusted list
ipv4 <IP of CM CPE> 255.255.255.255
ipv4 <IP of CM CPE> 255.255.255.255
ipv4 <IP of CM CPE> 255.255.255.255
ipv4 <IP of ITSP SBC> 255.255.255.255
ipv4 <IP of ITSP SBC> 255.255.255.255
rtp-port range 16384 32766
address-hiding
mode border-element
media anti-trombone
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
trace
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
header-passing
error-passthru
registrar server
no update-callerid
midcall-signaling preserve-codec
early-offer forced
midcall-signaling passthru
privacy-policy passthru
!
sip-ua no remote-party-id retry invite 2 timers trying 300 registrar ipv4:<IP of SBC inside interface> expires 3600 connection-reuse g729-annexb override


Response Signature


View solution in original post

26 Replies 26

moustafa.idc
Level 1
Level 1

Here is the debug output

MA-CME-01#
MA-CME-01#
*Nov 10 23:31:35.566: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB4E60C5A8 with refCount = 1
*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB61423870 with refCount = 1
*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/httpish_msg_process_network_msg: MSG LINE READ FAILURE DUE TO RS->EOF
*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/ccsip_process_network_message: process_network_msg: not complete
*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB61423870
*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads:
SIP Message incomplete, trashed
*Nov 10 23:31:35.567: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB4E60C5A8
*Nov 10 23:31:45.567: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB61421168 with refCount = 1
*Nov 10 23:31:45.567: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB616BBBE0 with refCount = 1
*Nov 10 23:31:45.567: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/httpish_msg_process_network_msg: MSG LINE READ FAILURE DUE TO RS->EOF
*Nov 10 23:31:45.567: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/ccsip_process_network_message: process_network_msg: not complete
*Nov 10 23:31:45.568: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB616BBBE0
*Nov 10 23:31:45.568: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


*Nov 10 23:31:45.568: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads:
SIP Message incomplete, trashed
*Nov 10 23:31:45.568: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB61421168
*Nov 10 23:31:46.950: //-1/xxxxxxxxxxxx/SIP/Info/verbose/4096/sipSPIAddContextToTable: Added context(0x7FFB5F4980A0) with key=[265] to table
*Nov 10 23:31:46.951: //-1/000000000000/SIP/Info/info/4096/ccsip_ipip_media_service_init:
*Nov 10 23:31:46.951: //-1/000000000000/SIP/Info/info/4096/ccsip_tdmip_media_service_init:
*Nov 10 23:31:46.951: //-1/000000000000/SIP/Info/verbose/36864/ccsip_ipip_media_forking_init: MF: Queue is initialised..
*Nov 10 23:31:46.951: //-1/000000000000/SIP/Info/verbose/4097/ccsip_platform_init_ccb: Initialising rtp session queue
*Nov 10 23:31:46.951: //264/000000000000/SIP/State/sipSPIChangeState: 0x7FFB5F4980A0 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Nov 10 23:31:46.951: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.951: //264/000000000000/SIP/Info/notify/131072/ccsip_call_setup_request: vrfID[0]
*Nov 10 23:31:46.951: //264/000000000000/SIP/Info/critical/32768/ccsip_ipip_media_forking_read_from_TDContainer: MF: Unable to read data from TD Container..
*Nov 10 23:31:46.951: //264/000000000000/SIP/Info/critical/32768/ccsip_ipip_media_forking_forked_leg_config: MF: TD container cannot be read/container is NULL. Setting of forked call leg failed..
*Nov 10 23:31:46.951: //264/000000000000/SIP/Info/verbose/4096/ccsip_iwf_map_ccapi_event_to_iwf_event: Event Category: 1, Event Id: EV_UNDEFINED
*Nov 10 23:31:46.951: //264/000000000000/SIP/Info/verbose/4096/ccsip_iwf_map_ccapi_event_to_iwf_event: IWF Event: E_SIP_IWF_EV_SET_MODE
*Nov 10 23:31:46.951: //264/000000000000/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 10 23:31:46.951: //264/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container
*Nov 10 23:31:46.951: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_sip_md_snr:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_sip_ed_snr:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_sip_md:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_sip_ed:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_h32x_in_set_mode:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_h323_in_set_mode:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/is_mode_sip_sccp_in_set_mode:
*Nov 10 23:31:46.952: //264/000000000000/SIP/Info/info/4096/ccsip_get_int_type_frm_set_mode_ev:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/is_mode_sip_sccp_in_set_mode:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/8192/sip_iwf_def_set_mode_hdlr: Setting SPI mode to SIP-TDM
*Nov 10 23:31:46.953: //264/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_sccp_early_dialog_container
*Nov 10 23:31:46.953: //264/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:next_state:CNFSM_NO_STATE_CHANGE
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/ccsip_iwf_handle_peer_event: Return value: SIP_IWF_SUCCESS
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/verbose/4096/ccsip_iwf_map_ccapi_event_to_iwf_event: Event Category: 3, Event Id: CC_EV_IF_DIAG_DONE
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/verbose/4096/ccsip_iwf_map_ccapi_event_to_iwf_event: IWF Event: E_SIP_IWF_EV_SET_FLOW_MODE
*Nov 10 23:31:46.953: //264/000000000000/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 10 23:31:46.953: //264/000000000000/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/is_fa2ft_md_flow_mode_transition:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/is_fa2ft_flow_mode_transition:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/ccsip_get_flow_mode_frm_set_flow_mode_ev:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/is_fa2ft_flow_mode_transition:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/ccsip_get_flow_mode_frm_set_flow_mode_ev:
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/verbose/4096/ccsip_iwf_process_event: IWF - cnfsm ret 2
*Nov 10 23:31:46.953: //264/000000000000/SIP/Info/info/4096/ccsip_iwf_handle_peer_event: Return value: SIP_IWF_SUCCESS
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/info/4096/ccsip_call_setup_request: Before processing SETUP REQccb->pld.flags_ipip = 200
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/info/4096/ccsip_call_setup_request: After processing SETUP REQccb->pld.flags_ipip = 200
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/critical/32768/ccsip_call_setup_request: MF or SIP TDM call flow
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/info/8192/sipSPIGenerateSessionUUID: Generate uuid
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/info/8192/sipSPIGenerateSessionUUID: Initial Invite SIP-TDM /MF case
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/info/8192/sipSPIGenerateSessionUUID: generated uuid - 8688eef22b345cf7b73e086c1cef835f
*Nov 10 23:31:46.954: //264/000000000000/SIP/Info/notify/8192/sipSPI_Add_SessionIDtoTDContainer: localUUID - 8688eef22b345cf7b73e086c1cef835fremoteUUID - 00000000000000000000000000000000
*Nov 10 23:31:46.954: //264/3104FDF580B3/SIP/Info/verbose/4096/ccsip_call_setup_request: Number Translation Set For Called-Number
*Nov 10 23:31:46.954: //264/3104FDF580B3/SIP/Info/info/2048/sipSPIGetExtensionCfg: SIP extension config:1, check sys cfg:1
*Nov 10 23:31:46.954: //264/3104FDF580B3/SIP/Info/verbose/5120/ccsip_call_setup_request: Session target or outbound proxy configured
*Nov 10 23:31:46.954: //-1/xxxxxxxxxxxx/SIP/Info/verbose/5120/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : sip.bluetelecoms.com target_port : 5060

*Nov 10 23:31:46.954: //264/3104FDF580B3/SIP/Info/verbose/4096/sipSPIUaddccCallIdToTable: Adding call id 108 to table
*Nov 10 23:31:46.954: //264/3104FDF580B3/SIP/Info/notify/131072/ccsip_call_setup_request: Incrementing call counter to [1] in dial-peer [100]
*Nov 10 23:31:46.955: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Nov 10 23:31:46.956: //-1/xxxxxxxxxxxx/SIP/Info/info/4096/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 2
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/2049/sipSPIGetCallConfig: preferred_codec set[0] type :No Codec bytes: 0
*Nov 10 23:31:46.956: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/notify/131072/sipSPIGetCallConfig: peer_tag = 100, tenant_tag = 0, VRFId = 0
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/2048/sipSPISetAudioForcedConfig: audio forced config is set to FALSE
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/verbose/2048/sipSPI_ipip_store_config_info: Setting mid_call_config_info = 0x0 for callid = 264
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/8192/sipSPIGetCallConfig: Media forking disabled
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/32768/sipSPIGetCallConfig: Media Antitrombone disabled
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/notify/131072/sipSPICanSetFallbackFlag: Local Fallback is not active
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/8192/sipSPIGetCallConfig: VRF id = 0
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/8192/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
*Nov 10 23:31:46.956: //264/3104FDF580B3/SIP/Info/info/2304/sipSPISetMediaFlowMode: xcoder high-density disabled
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/8192/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/1/sipSPIGetCallConfig: Using Voice Class Codec, tag = 1 and offer-all is = FALSE
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: STUN Usage is not enabled
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/verbose/12288/sipSPIGetModemInfoPerCall: peer_callID=263
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/critical/32768/ccsip_ipip_media_forking_update_preferred_codec: MF: Not a Forked SIP leg..
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/8704/sipSPIGetCallConfig: Incoming: No defer BYE for last
call stats
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/verbose/1/ccsip_set_srtp_config: No Srtp configure for this leg.
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/8192/sipSPIGetCallConfig: Media forking disabled
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/34816/ccsip_ipip_media_forking_anchor_leg_config: MF: en_p->encap_s.voIP.voipPeerCfgMediaClass = 0
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/34816/ccsip_ipip_media_forking_anchor_leg_config: MF: Dial-peer has no media class recorder.
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/36864/sipSPIMFChangeState: MF: Prev state = 0 & New state = -1
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/32768/ccsip_ipip_media_forking_anchor_leg_reset: MF: Anchor leg config reset done...
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/32768/ccsip_ipip_media_forking_intra_frame_request_config: MF: FIR en_p->encap_s.voIP.voipPeerCfgMediaClass = 0
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/info/32768/ccsip_ipip_media_forking_get_forked_leg_config: MF: This leg is not forked call leg.
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/critical/11264/ccsipInitDSCPPolicyInfo: No DSCP Profile configured, No RPH 2 DSCP Mapping and DSCP policing
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/verbose/8192/sipSPIGetCallConfig: Initilise the DSCP policy
*Nov 10 23:31:46.957: //264/3104FDF580B3/SIP/Info/verbose/8192/sipSPICheckFAAnatAssymetricOrDO2EO: Not a SIP-SIP call or not in FA mode
*Nov 10 23:31:46.958: //-1/xxxxxxxxxxxx/SIP/Info/info/64/sipSPISetSipProfilesTag: voice class SIPProfiles tag is set : 1
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/notify/2049/populate_vcc_data: Using Voice Class Codec, tag = 1 and offer-all is = FALSE
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/notify/8192/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/verbose/10240/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/verbose/2048/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/info/2048/sipSPIGetExtensionCfg: SIP extension config:1, check sys cfg:1
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/notify/10240/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/verbose/5120/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/info/1/preprocessSetup:
This is a not a SIGO Call -, could be DM call
*Nov 10 23:31:46.958: //264/3104FDF580B3/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 10 23:31:46.958: //264/3104FDF580B3/SIP/State/ccsip_cnfsm_debugs: IWF:new_container:sip_iwf_main_container
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/verbose/4096/ccsip_iwf_process_event: IWF - cnfsm ret 2
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/notify/4096/preprocessSetup: SIP-TDM or TCL/VXML app case
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/notify/4096/sip_gw_pre_setup_update_stream_media_direction: peer_callID = 263
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/critical/4097/sip_gw_pre_setup_update_stream_media_direction: peer_channels/stream is NULL
*Nov 10 23:31:46.958: //264/3104FDF580B3/SIP/Info/notify/1/sip_gw_pre_setup_add_sdp_container: DNS/ENUM resolution required; Deferred Creating SDP
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/131072/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 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/131072/sipSPIBwCacUpdateInterfaceBw: bwcac acquiring interface GigabitEthernet0/0/0 bw 0
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/131072/sipSPIBwCacUpdateAccountedBw: bwcac update accounted bw (initial offer) accounted bw set to 0
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/133120/sipSPIBwCacIsDialPeerBwAvailable: bwcac NOP dial-peer bw available tag 100
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/notify/1/sipSPIBwCacIsInterfaceBwAvailable: bwcac interface bw threshold not configured
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/131072/sipSPIBwCacVerifyBwThreshold: bwcac verify bw threshold, bw available allow call total bw 0 bps
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/critical/8192/sipSPIValidateGtd: Signal Forward disabled
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/critical/8192/sipSPIValidateTunnelData: RawMsg/QSIG Tunneling Not Enabled
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/critical/10240/sipSPIAddMLPPServicesInfo: No MLP Info available on incoming leg
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/3072/sipSPIPreprocessUriFormat: Url cfg for 1: 2,phone-ctxt=FALSE
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/info/9216/sipSPIAddCiscoGcid: Gcid value not set - not adding header.
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/verbose/4096/sipSPIAddPrivacyandIdentityInfo: ccb->local_host_name,ccb->src_addr_str is NULL
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Info/critical/2048/sipSPICompareHistoryInfoWithMatchedDialpeer: call-route history-info CLI not enabled
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/Error/sipSPI_ipip_set_history_info_header:
ccb->src_addr_str is NULL
*Nov 10 23:31:46.959: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE
*Nov 10 23:31:46.959: //264/3104FDF580B3/SIP/State/sipSPIChangeState: 0x7FFB5F4980A0 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_SENT_DNS)
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/State/sipSPIChangeState: 0x7FFB5F4980A0 : State change from (STATE_IDLE, SUBSTATE_SENT_DNS) to (STATE_IDLE, SUBSTATE_SENT_DNS)
*Nov 10 23:31:46.960: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.960: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.960: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: signaling bind address : 37.236.146.168
*Nov 10 23:31:46.960: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: bind_inuse: 1conn_reuse: 0
*Nov 10 23:31:46.960: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: return addr 37.236.146.168
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/Info/verbose/4096/sipSPIUaddCcbToUACTable: ****Adding to UAC table.0x7FFB5F4980A0 34A296BA-41B511EC-80B894FB-C2215BD2@sip.bluetelecoms.com
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/Info/verbose/4096/sipSPIUaddCcbToTable: Added to table. ccb=0x7FFB5F4980A0 key=34A296BA-41B511EC-80B894FB-C2215BD2@sip.bluetelecoms.com balance 1
*Nov 10 23:31:46.960: //264/3104FDF580B3/SIP/Info/verbose/12288/sipSPIUsetBillingProfile: sipCallId for billing records = 34A296BA-41B511EC-80B894FB-C2215BD2@sip.bluetelecoms.com
*Nov 10 23:31:46.961: //-1/xxxxxxxxxxxx/SIP/Info/notify/8192/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.sip.bluetelecoms.com and type:1
*Nov 10 23:31:46.964: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/sip_dns_type_a_aaaa_query: DNS query for sip.bluetelecoms.com and type:1
*Nov 10 23:31:46.966: //-1/xxxxxxxxxxxx/SIP/Info/notify/8192/sip_dns_type_a_query: TYPE A query successful for sip.bluetelecoms.com
*Nov 10 23:31:46.966: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/sip_dns_type_a_query: ttl for A records = 146 seconds
*Nov 10 23:31:46.966: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/sip_dns_type_a_aaaa_query: IP Address of sip.bluetelecoms.com is:

*Nov 10 23:31:46.966: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/sip_dns_type_a_aaaa_query: 185.32.74.8

*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/4096/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 43
*Nov 10 23:31:46.967: //264/3104FDF580B3/SIP/State/sipSPIChangeState: 0x7FFB5F4980A0 : State change from (STATE_IDLE, SUBSTATE_SENT_DNS) to (STATE_IDLE, SUBSTATE_NONE)
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/4096/sipSPICacheHostToCCB: sipSPICacheHostToCCB dnsResponse.num_hosts = 1
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/4096/sipSPICacheHostToCCB: IP Address No. 1, IP address 185.32.74.8
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.967: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.967: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.967: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: signaling bind address : 37.236.146.168
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: bind_inuse: 1conn_reuse: 0
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: return addr 37.236.146.168
*Nov 10 23:31:46.967: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.967: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.967: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: signaling bind address : 37.236.146.168
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: bind_inuse: 1conn_reuse: 0
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: return addr 37.236.146.168
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/info/131072/sipSPIRscmsmAvail: Value returned by check is = 0
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/critical/2048/sipSPICompareHistoryInfoWithMatchedDialpeer: call-route history-info CLI not enabled
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/critical/1024/sipSPI_ipip_set_history_info_header: No HI header recvd from container
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/verbose/2048/sipSPIAddPrivacyandIdentityInfo: asserted-id is configured as pai header
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/notify/6/sipSPIValidateStreamAddrType: stream:1, Mode : 1
*Nov 10 23:31:46.968: //264/3104FDF580B3/SIP/Info/verbose/513/resolve_media_ip_address_to_bind: peer_tag=100
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_media_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.968: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.969: //264/3104FDF580B3/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 37.236.146.168
*Nov 10 23:31:46.969: //264/3104FDF580B3/SIP/Info/critical/1/sipSPIOutgoingCallSDP: Failure in creating outbound streams
SIP: (264) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 10 23:31:46.969: //-1/xxxxxxxxxxxx/SIP/Info/verbose/512/ccsip_get_vrf_from_sip_bind: vrfid 0 for tag 100, sip_vrfid 0, voice_vrfid 0
*Nov 10 23:31:46.969: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.969: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: VRF id = 0
*Nov 10 23:31:46.969: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/ccsip_get_ifaddress: ip_address IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.969: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_ip_address_to_bind: ip_get_ifaddress IPv4 37.236.146.168 for SIP
*Nov 10 23:31:46.969: //264/3104FDF580B3/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: signaling bind address : 37.236.146.168
*Nov 10 23:31:46.969: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: bind_inuse: 1conn_reuse: 0
*Nov 10 23:31:46.969: //-1/xxxxxxxxxxxx/SIP/Info/info/8192/resolve_sig_ip_address_to_bind: return addr 37.236.146.168
*Nov 10 23:31:46.969: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 8044 for stream 1
*Nov 10 23:31:46.969: //264/3104FDF580B3/SIP/Info/info/1/sipSPIDoBearerCapToCodecMapping: Bearer capability to Codec Mapping: DISABLED

*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/info/131074/sipSPIBwCacCalcMaxAudioBandwidth: calculating max bw from preffered codecs (local offer)
SIP: (264) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/info/131074/sipSPIBwCacCalcMaxAudioBandwidth: max bw (excluding pak overhead) from preffered codecs: codec g711alaw bw 64000 index 0
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/critical/2/sipSPIBwCacCalcMaxAudioBandwidth: audio caps channel idx not found !!!!
*Nov 10 23:31:46.970: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/info/131074/sipSPIBwCacCalcMaxAudioBandwidth: max bw (including pak overhead) from preffered codecs: codec g711alaw bw 80000
*Nov 10 23:31:46.970: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
*Nov 10 23:31:46.970: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
*Nov 10 23:31:46.970: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/convert_codec_bytes_to_ptime: Values :Codec: g729r8 codecbytes :20, ptime: 20
*Nov 10 23:31:46.970: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/convert_codec_bytes_to_ptime: Values :Codec: g729br8 codecbytes :20, ptime: 20
*Nov 10 23:31:46.970: //-1/xxxxxxxxxxxx/SIP/Info/notify/1/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/notify/8193/sip_generate_sdp_xcaps_list: Negotiation not done v150_capable 0
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/notify/8193/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Info/info/1/sipSPIOutgoingCallSDP: Creating recv-only stream for outbound call
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE
*Nov 10 23:31:46.970: //264/3104FDF580B3/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Info/info/131072/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 10 23:31:46.971: //264/3104FDF580B3/SIP/Info/info/139264/sipSPIBwCacUpdateInterfaceBw: NOP (no interface change)
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Info/info/131072/sipSPIBwCacUpdateAccountedBw: bwcac update accounted bw (no interface change)
*Nov 10 23:31:46.971: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 185.32.74.8,Port 5060, Transport 1, SentBy Port 5060vrfid 0
*Nov 10 23:31:46.971: //-1/xxxxxxxxxxxx/SIP/Info/verbose/8192/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Nov 10 23:31:46.971: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB453F8A38 with refCount = 1
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Info/info/8192/sipSPIGetTDSessionID: Get session-ID local uuid - 8688eef22b345cf7b73e086c1cef835f remote uuid - 00000000000000000000000000000000
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Info/notify/8192/sipSPIAddSessionID: localUUID - 8688eef22b345cf7b73e086c1cef835f remoteUUID - 00000000000000000000000000000000
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Info/notify/8192/sipSPIAddSessionID: Session-ID header 8688eef22b345cf7b73e086c1cef835f;remote=00000000000000000000000000000000
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
*Nov 10 23:31:46.971: //264/3104FDF580B3/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:7FFB616AC2A8
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/info/8192/Session-Timer/sipSTSLSRReqSend: Session timer is not required
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/info/8192/Session-Timer/sipSTSLMain:
SE: 0;refresher:none peer refresher:none, flags:2000, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:3600, Configured Min-SE:3600
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/verbose/8192/sipSPIPresendProcessing: Presend Processing called for 0 event
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/notify/4096/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/notify/4096/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer: Could not get any elements from TD Container
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/info/4096/ccsip_offer_ans_handle_sent_sdp:
*Nov 10 23:31:46.972: //264/3104FDF580B3/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 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/info/4096/ccsip_offer_ans_is_invite_offer_valid: TRUE
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/info/4096/ccsip_offer_ans_common_offer_sent_hdlr:
*Nov 10 23:31:46.972: //264/3104FDF580B3/SIP/Info/info/4096/ccsip_iwf_handle_network_event:
*Nov 10 23:31:46.972: //264/3104FDF580B3/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 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/info/4096/is_sent_sccp_do_video_inactive:
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/info/4096/is_mode_sip_sccp_do_video:
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/info/4096/is_mode_sip_sccp_do_video:
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/info/4096/sip_iwf_def_ed_sent_sdp_offer_hdlr:
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/State/ccsip_cnfsm_debugs: IWF:next_state:S_SIP_IWF_SDP_SENT_AWAIT_SDP
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/State/ccsip_cnfsm_debugs: OA:next_state:S_SIP_EARLY_DIALOG_OFFER_SENT
*Nov 10 23:31:46.973: //-1/xxxxxxxxxxxx/SIP/Info/info/2048/sipSPIgetRegistrarHost: registrar host retrieved : sip.bluetelecoms.com
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/critical/4096/ccsip_ipip_media_forking_get_forked_recording_data: MF: Not a Forked leg..
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/critical/1024/sipSPICreateRecParticipantHeaders: X-Cisco-Recording-Participant header not added.
SIP: (264) Group (a= group line) attribute, level 65535 instance 1 not found.
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/info/34816/sipSPIGetCallExtensionSupported: anat enabled, src_sdp dont have anat
*Nov 10 23:31:46.973: //264/3104FDF580B3/SIP/Info/info/4096/sipSPISendInvite: Associated container=0x7FFB616AC2A8 to Invite
*Nov 10 23:31:46.974: //-1/xxxxxxxxxxxx/SIP/Info/verbose/8192/sipSPIAppHandleContainerBody: sipSPIAppHandleContainerBody len 324
*Nov 10 23:31:46.974: //264/3104FDF580B3/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
*Nov 10 23:31:46.974: //264/3104FDF580B3/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Dial peer configuration, Switch Transport is FALSE
*Nov 10 23:31:46.974: //264/3104FDF580B3/SIP/Transport/sipSPITransportSendMessage: msg=0x7FFB453F8A38, addr=185.32.74.8, port=5060, sentBy_port=0, local_addr=37.236.146.168, is_req=1, transport=1, switch=0, callBack=0x55A85BE68CB0ACK sip:12146043977@sip.bluetelecoms.com:5060 SIP/2.0
Via: SIP/2.0/UDP 37.236.146.168:5060;branch=z9hG4bK133FD9
From: "SIP" <sip:442037752050@sip.bluetelecoms.com>;tag=9FD5FA-5CA
To: <sip:12146043977@sip.bluetelecoms.com>;tag=860yD6tcKyg0e
Date: Wed, 10 Nov 2021 23:31:46 GMT
Call-ID: 34A296BA-41B511EC-80B894FB-C2215BD2@sip.bluetelecoms.com
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 00000000000000000000000000000000;remote=8688eef22b345cf7b73e086c1cef835f
Content-Length: 0


*Nov 10 23:31:47.187: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB453F8A38
MA-CME-01#
MA-CME-01#
MA-CME-01#ter
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB4E60C5A8 with refCount = 1
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_create: created msg=0x7FFB4E431A30 with refCount = 1
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/httpish_msg_process_network_msg: MSG LINE READ FAILURE DUE TO RS->EOF
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Info/critical/1024/ccsip_process_network_message: process_network_msg: not complete
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB4E431A30 no m
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads:
SIP Message incomplete, trashed
*Nov 10 23:31:55.568: //-1/xxxxxxxxxxxx/SIP/Info/info/1024/httpish_msg_free: Freed msg=0x7FFB4E60C5A8on
MA-CME-01#und
MA-CME-01#undebug all
MA-CME-01#undebug all
All possible debugging has been turned off
MA-CME-01#
MA-CME-01#
MA-CME-01#

 

Leonardo Santana
Spotlight
Spotlight

Hi,

Could your attach a debug ccsip message?

All calls are with this isssue?

Inbound calls are good?

Regards

Leonardo Santana

Regards
Leonardo Santana

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outbound is the only issue, when placing a call from ip phone through sip trunk I have busy tone

 

Here is the debug,

 

*Nov 11 01:00:54.114: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


*Nov 11 01:00:59.511: //385/A7390B4B80C3/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:12146043977@sip.bluetelecoms.com:5060 SIP/2.0
Via: SIP/2.0/UDP 37.236.146.168:5060;branch=z9hG4bK1A62026
From: "SIP" <sip:442037752050@37.236.146.168>;tag=F18253-D77
To: <sip:12146043977@sip.bluetelecoms.com>
Date: Thu, 11 Nov 2021 01:00:59 GMT
Call-ID: AAE6F7EB-41C111EC-80C894FB-C2215BD2@sip.bluetelecoms.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 3600
Cisco-Guid: 2805533515-1103172076-2160301307-3256966098
User-Agent: Cisco-SIPGateway/IOS-16.6.4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1636592459
Contact: <sip:442037752050@37.236.146.168:5060>
Expires: 1800
Allow-Events: telephone-event
P-Asserted-Identity: "SIP" <sip:442037752050@37.236.146.168>
Session-ID: 90098c2c84235ced90840aa8b1a2ac3c;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 324

v=0
o=CiscoSystemsSIP-GW-UserAgent 6598 8374 IN IP4 37.236.146.168
s=SIP Call
c=IN IP4 37.236.146.168
t=0 0
m=audio 8048 RTP/AVP 8 0 18 101
c=IN IP4 37.236.146.168
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Nov 11 01:00:59.621: //385/A7390B4B80C3/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 37.236.146.168:5060;branch=z9hG4bK1A62026
From: "SIP" <sip:442037752050@37.236.146.168>;tag=F18253-D77
To: <sip:12146043977@sip.bluetelecoms.com>
Call-ID: AAE6F7EB-41C111EC-80C894FB-C2215BD2@sip.bluetelecoms.com
CSeq: 101 INVITE
server: SBC
Content-Length: 0


*Nov 11 01:00:59.792: //385/A7390B4B80C3/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 37.236.146.168:5060;branch=z9hG4bK1A62026
Max-Forwards: 30
From: "SIP" <sip:442037752050@37.236.146.168>;tag=F18253-D77
To: <sip:12146043977@sip.bluetelecoms.com>;tag=t80Q1SaSttFZa
Call-ID: AAE6F7EB-41C111EC-80C894FB-C2215BD2@sip.bluetelecoms.com
CSeq: 101 INVITE
User-Agent: bluetelecoms
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0


*Nov 11 01:00:59.795: //385/A7390B4B80C3/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:12146043977@sip.bluetelecoms.com:5060 SIP/2.0
Via: SIP/2.0/UDP 37.236.146.168:5060;branch=z9hG4bK1A62026
From: "SIP" <sip:442037752050@sip.bluetelecoms.com>;tag=F18253-D77
To: <sip:12146043977@sip.bluetelecoms.com>;tag=t80Q1SaSttFZa
Date: Thu, 11 Nov 2021 01:00:59 GMT
Call-ID: AAE6F7EB-41C111EC-80C894FB-C2215BD2@sip.bluetelecoms.com
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 00000000000000000000000000000000;remote=90098c2c84235ced90840aa8b1a2ac3c
Content-Length: 0


*Nov 11 01:01:04.115: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:


*Nov 11 01:01:14.115: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

 

Hi,

For some reason your receiving a 404 not found from your ITSP:

image.png

The dialplan is correct?

This is a new implementation?

Did you contact your ITSP?

Regards

Leonardo Santana

Regards
Leonardo Santana

*** Rate All Helpful Responses***

moustafa.idc
Level 1
Level 1

DialPlan is correct, as this was working before adding the new SIP phones. I contacted ITSP earlier and they said they have not change anything at their end. incoming calls through out same ITSP work just fine. not sure what is the deal.

Moustafa,

The ITSP is returning a 404 not found, i think you need to check with them why they are sending your this disconnection reason.

Collect these ouputs too
show sip-ua register status
debug voip ccapi inout - (make the call)

Regards

Leonardo Santana

 

Regards
Leonardo Santana

*** Rate All Helpful Responses***

Could you please share your configuration ?

 

 



Response Signature


Hi Nithin,

 

here is the config file

Check with your ISP, why they respond with 404.

Mention the below details when opening a ticket with your ISP.

 

Received:
SIP/2.0 404 Not Found



Response Signature


Hi Nithin,

 

I contacted ITSP and they still investigate the issue.

My only concern is why the FROM is showing my CME IP in the 404 NOT found message? I think it should be the other way.

I might change the sip profile rules and see.

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

*Nov 11 19:35:14.148: //1914/4FB46F0481EC/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 37.236.146.168:5060;branch=z9hG4bK7451AEE
Max-Forwards: 30
From: "SIP" <sip:442037752050@37.236.146.168>;tag=4EDA064-232A
To: <sip:12146043977@sip.bluetelecoms.com>;tag=yj28cUma2HQcF
Call-ID: 535674CF-425D11EC-81F194FB-C2215BD2@sip.bluetelecoms.com
CSeq: 101 INVITE
User-Agent: bluetelecoms
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

Try to move the ITSP registrar bit to a tenant configuration. This would be an example of how that could look like, you'd need to alter it to fit your specifics.

voice class tenant 2000
  registrar dns:<service provider DNS name> expires 3600
  credentials username <user name for connection> password 7 <password for connection> realm <service provider realm name>
  authentication username <user name for connection> password 7 <password for connection>
  no remote-party-id
  timers dns registrar-cache 95
  sip-server dns:<service provider DNS name>
  connection-reuse
  audio forced
  bind control source-interface GigabitEthernet0/0/1
  bind media source-interface GigabitEthernet0/0/1
  no pass-thru content custom-sdp
  sip-profiles 10
  outbound-proxy dns:<service provider DNS name for proxy> reuse
  early-offer forced


dial-peer voice 100 voip
 description Inbound calls from PSTN
 translation-profile incoming PSTN-IN
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 2000  
 voice-class sip tenant 2000
 dtmf-relay rtp-nte
 no vad   
!
dial-peer voice 110 voip
 description Outbound calls to PSTN
 translation-profile outgoing PSTN-OUT
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 2000  
 voice-class sip tenant 2000
 voice-class sip options-keepalive profile 2000
 dtmf-relay rtp-nte
 no vad

The reason I think this is affecting you is that SIP phones need to have a registrar under sip-ua, so it interference with what you have for the ITSP.

 

This is from a SBC where we run a SIP connection that uses registration and we're having SRST for SIP phones on it as well. As SRST and CME is pretty much the same thing when it comes to this part it should fit.

voice service voip
ip address trusted list
ipv4 <IP of CM CPE> 255.255.255.255
ipv4 <IP of CM CPE> 255.255.255.255
ipv4 <IP of CM CPE> 255.255.255.255
ipv4 <IP of ITSP SBC> 255.255.255.255
ipv4 <IP of ITSP SBC> 255.255.255.255
rtp-port range 16384 32766
address-hiding
mode border-element
media anti-trombone
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
trace
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
header-passing
error-passthru
registrar server
no update-callerid
midcall-signaling preserve-codec
early-offer forced
midcall-signaling passthru
privacy-policy passthru
!
sip-ua no remote-party-id retry invite 2 timers trying 300 registrar ipv4:<IP of SBC inside interface> expires 3600 connection-reuse g729-annexb override


Response Signature


Thanks a lot Roger, this is exactly what I was seeing.

What I did I remove the SIP phones and get the SIP trunk call back to work. then I have applied Voice Class Tenant. and change sip-ua config as per your recommendations.

I have now successful call to the ITSP (incoming and outgoing) still need to figure out away to have dns name instead of IPv4:

 

voice class server-group 2000

ipv4 185.32.XX.XX

 

dial-peer voice 100 voip

session server-group 2000

 

once I have this step done, I will configure the sip phones and let you know the result.

I really appreciate the time and effort.

With DNS name you would remove the server group and put the DNS in a regular on dial peer configuration.

Please see this document for details. https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html



Response Signature


And very happy to hear that you got the calls to work.



Response Signature