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CUBE configuration with CUCM Call Manager

sarwarm123
Level 1
Level 1

Our company recently decided to have a SIP trunk in our Cisco VOIP environment. I have not configured CUBE router before so I have read number of documents related to CUBE configuration with CUCM. However I still have confusion so I need your help. Please see proposed network diagram below.

SIP trunk.JPG

My Question is, do we connect one gig port to company LAN where CUCM is connected and second gig interface will connect to SIP Provider customer end router? For example

interface GigabitEthernet0/0

description Connection to LAN

ip address 172.102.243.10 255.255.255.0

duplex auto

speed auto

!

interface GigabitEthernet0/1

description connection to SIP Provider

ip address 99.102.153.24 255.255.255.0

duplex auto

speed auto

I have seen some other sample configurations where people configured only one gig port for example (I have also attached full sample configuration)

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

no ip address

shutdown

duplex auto

speed auto

service-policy output VOIP-Policy

!

interface GigabitEthernet0/1

description $ES_LAN$

ip address 146.191.201.41 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 146.191.201.41

service-policy output VOIP-Policy

65 Replies 65

Hi Aok,

Thanks for your help. Using command "no remote-party id" under sip-ua was the typing a mistake which I have corrected. However I have applied config on the CUBE router but experiencing an issue.

ITSP has provided us a test number so for testing I have created a translation pattern in CUCM to translate 2476012665 (test number) to 5 digit VOIP number 64048. Unfortunately my initial test of incoming call is not working. I have enabled following debug but not seeing any messages or errors either.

Please note: I am using SCCP phone to receive incoming call from ITSP

CDC-CUBE# sh debugging

VOIP RTP ALL Named Events debugging is on

CCAPI:

debug voip ccapi inout is ON (filter is OFF)

CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)

CCSIP SPI: SIP Call Message tracing is enabled (filter is OFF)

CCSIP SPI: SIP Call State Machine tracing is enabled   (filter is OFF)

CCSIP SPI: SIP Call Events tracing is enabled   (filter is OFF)

CCSIP SPI: SIP error debug tracing is enabled   (filter is OFF)

CCSIP SPI: SIP info debug tracing is enabled   (filter is OFF)

CCSIP SPI: SIP media debug tracing is enabled   (filter is OFF)

CCSIP SPI: SIP Call preauth tracing is enabled (filter is OFF)

CCSIP SPI: SIP Call transport tracing is enabled       (filter is OFF)

CCSIP SPI: SIP DHCP tracing is enabled (filter is OFF)

Muhammad,

First of all how are you looking at the logs? Is your gateway configured to send logs to the buffer? If it is..then you need to use the command "show loggin" to see the logs...If you are loggin into conslole and you telnet into the gateway then you need to enablle "term mon"

If you have done all of this and you dont have any logs, then the call is not coming into your gateway. You need to go back and speak to your provider

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi Aok,

Hope you are fine. I can see now logs, however incoming calls are not working and getting follwoing error messages. I am doing following debug

CCAPI:

  debug voip ccapi inout is ON (filter is OFF)

CCSIP SPI: SIP Call Message tracing is enabled  (filter is OFF)

First, If I look the logs I dont see any matching dial-peer second seeing this error SIP/2.0 404 Not Found

Full log file has attached

So I can see the called number is 02476012665..

The inbound dial-peer that is matched shows dial-peer 0 and no outbound dial-peer matched

Looking at your inbound dial-peer from ITSP it has

incoming called number [2-9]......... (which does not have any 0 infront of it)

Please change the config to

dial-peer voice 1 voip

description *** Inbound WAN side dial-peer ***

session protocol sipv2

incoming called-number 0[2-9].........

dtmf-relay rtp-nte

codec g711ulaw

and dial-peer 20 and 21 to

dial-peer voice 20 voip

description *** Outbound LAN side dial-peer ***

destination-pattern 0[2-9].........

session protocol sipv2

session target ipv4:10.206.143.13

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 21 voip

description *** Outbound LAN side dial-peer ***

destination-pattern 0[2-9].........

session protocol sipv2

session target ipv4:10.206.143.12

dtmf-relay rtp-nte

codec g711ulaw

Ensure that the CSS on your sip trunk has the partition of the xlation pattern you have configured to change the called number to your internal number

Test again and send the debugs

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi mate,

Just a addition, after changing the incoming dial-peer to this

dial-peer voice 101 voip

description ***Inbound calls from CUCM****

translation-profile incoming SIP-TRUNK-IN

session protocol sipv2

incoming called-number 02476012665

dtmf-relay rtp-nte

codec g711ulaw

I am getting this error message 488 Media Not Acceptable. Full log have attached

OK,

You rprovider is advertising only g711alaw

Chnage your config to

dial-peer voice 101 voip

codec g711alaw

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi Aok,

As you said I have updated codec but getting this error now. Full log has attached

Muhammad,

do the following:

send the updated config on dial-peer 20, 22 and 23

then do another test call and send me only

debug ccsip all

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi Aok,

As you said I have attached debug file and please see below dial-peer config

dial-peer voice 101 voip
description *** Inbound Calls from ITSP ***
translation-profile incoming SIP-TRUNK-IN
session protocol sipv2
incoming called-number 02476012665
dtmf-relay rtp-nte
codec g711alaw
!
dial-peer voice 20 voip
description **** Outbound calls to Primary CUCM ***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:10.102.243.13
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 21 voip
description *** Outbound Calls to Backup CUCM***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:10.102.243.12
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 22 voip
description *** Outbound LAN side dial-peer ***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:192.168.132.2
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 23 voip
description *** Outbound LAN side dial-peer ***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:192.168.162.253
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 1 voip
description *** Inbound Calls from ITSP ***
session protocol sipv2
incoming called-number [2-9].........
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 2 voip
description *** Outbound calls to ITSP ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw

User Access Verification

Username: admin
Password:
CDC-CUBE#debug ccsip all
This may severely impact system performance. Continue? [confirm]
All SIP Call tracing is enabled
CDC-CUBE#
CDC-CUBE#term mon
CDC-CUBE#
009786: Mar 26 17:33:38.818: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
009787: Mar 26 17:33:38.818: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
009788: Mar 26 17:33:38.818: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x264956C, addr=192.168.132.3, port=36609, local_addr=, connid=2, transport=TCP
009789: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/sipHolderGetConnInstance: Return existing connection for port 36609 connId 2
009790: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.116.143.151:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.132.3:5060;branch=z9hG4bK60e4736922fb
From: <192.168.132.3>;tag=1872667285

CDC-CUBE#To: <10.116.143.151>
Date: Tue, 26 Mar 2013 17:33:38 GMT
Call-ID: 49c70a80-1511dbf2-2b74-384a8c0@192.168.132.3
User-Agent: Cisco-CUCM8.6
CSeq: 101 OPTIONS
Contact: <192.168.132.3:5060>
Max-Forwards: 0
Content-Length: 0


009791: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
009792: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x2D13478) with key=[917] to table
009793: Mar 26 17:33:38.822: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
009794: Mar 26 17:33:38.822: //-1/000000000000/SIP/Info/ccsip_iwf_init:
009795: Mar 26 17:33:38.822: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
009796: Mar 26 17:33:38.822: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
009797: Mar 26 17:33:38.822: //-1/000000000000/SIP/Info/ccsip_ipip_media_forking_init: MF: Queue is initialised..
009798: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.132.3,Port 36609, Transport 2, SentBy Port 5060
009799: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009800: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009801: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.116.143.151
009802: Mar 26 17:33:38.822: //-1/2112D08B8484/SIP/State/sipSPIChangeState: 0x2D13478 : State ch
CDC-CUBE#ange from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
009803: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.132.3,Port 36609, Transport 2, SentBy Port 5060
009804: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
009805: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.132.3,Port 36609, Transport 2, SentBy Port 5060
009806: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetViaHostInURLFormat: VIA URL Conversion failed for
009807: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: :
009808: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Error/sipSPIGetPeerByCalledPartyId:
input arg error
009809: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
009810: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Error/sipSPIUpdateCallInfo:
input argument error
009811: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
009812: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
009813: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
009814: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
009815: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
009816: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
009817: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
009818: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
009819: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
009820: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 9
009821: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
009822: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
009823: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
009824: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 255
009825: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
009826: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0
009827: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Error/ccsip_ipip_media_forking_anchor_leg_config:
MF: Dial-peer is absent..
009828: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIMFChangeStat
CDC-CUBE#e: MF: Prev state = 0 & New state = -1
009829: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/ccsip_ipip_media_forking_anchor_leg_reset: MF: Anchor leg config reset done...
009830: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: Non dial peer leg - using RTP Supported Codecs
009831: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 18
009832: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 0
009833: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 8
009834: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 9
009835: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 4
009836: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 2
009837: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 15
009838: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 255
009839: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
009840: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
009841: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
009842: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled
009843: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
009844: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009845: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009846: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: return addr 10.116.143.151
009847: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.116.143.151
009848: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009849: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009850: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.116.143.151
009851: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_OPTIONS_RESP
009852: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sact_idle_new_message_options: ccsip_api_options_ind returned: SIP_SUCCESS
009853: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/State/sipSPIChangeState: 0x2D13478 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OPTIONS_WAIT, SUBSTATE_NONE)
009854: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
009855: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x2D13478 key=49c70a80-1511dbf2-2b74-384a8c0@192.168.132.3
009856: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 395 to table
009857: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 38
009858: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPISendOptionsResponse: Associated container=0x13D05388 to Options Response
009859: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Transport/sipSPITransportSendMessage: msg=0x2DAFD1C, addr=192.168.132.3, port=36609, sentBy_port=5060, local_addr=, is_req=0, transport=2, switch=0, callBack=0x88EBB00
009860: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
009861: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
009862: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x2DAFD1C, addr=192.168.132.3, port=36609, local_addr=, connId=2 for TCP
009863: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 395
009864: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[917] removed.
009865: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.
009866: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x2D13478 key=49c70a80-1511dbf2-2b74-384a8c0@192.168.132.3
009867: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.
009868: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
009869: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerUnregisterCtxtInConnection: gConnTab=0x264956C, addr=192.168.132.3, port=36609, local_addr=, unregistering context=0x2D13478
009870: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Transport/sipSPITransportContextCleanup: Could not purge context gcb=0x2D13478 from the connection; gcb might be locked
009871: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
009872: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/ccsip_offer_ans_delete:
009873: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/ccsip_iwf_delete:
009874: Mar 26 17:33:38.822: //917/2112D08B8484/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 0x2D13478
009875: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[917]
009876: Mar 26 17:33:38.822: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPITransportSendMessage: gcb has cleanedup, Done with send operations for this context
009877: Mar 26 17:33:38.826: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[917]
009878: Mar 26 17:33
CDC-CUBE#:38.826: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.132.3:5060;branch=z9hG4bK60e4736922fb
From: <192.168.132.3>;tag=1872667285
To: <10.116.143.151>;tag=44F9530-24F1
Date: Tue, 26 Mar 2013 17:33:38 GMT
Call-ID: 49c70a80-1511dbf2-2b74-384a8c0@192.168.132.3
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 379

v=0
o=CiscoSystemsSIP-GW-UserAgent 2932 5209 IN IP4 10.116.143.151
s=SIP Call
c=IN IP4 10.116.143.151
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.116.143.151
m=image 0 udptl t38
c=IN IP4 10.116.143.151
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

009879: Mar 26 17:33:38.906: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
009880: Mar 26 17:33:38.906: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
009881: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x264956C, addr=192.168.162.253, port=42653, local_addr=, connid=3, transport=TCP
009882: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/sipHolderGetConnInstance: Return existing connection for port 42653 connId 3
009883: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.116.143.151:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.162.253:5060;branch=z9hG4bK30e8c572db0bb
From: <192.168.162.253>;tag=1151533917
To: <10.116.143.151>
Date: Tue, 26 Mar 2013 17:33:38 GMT
Call-ID: 49c70a80-1511dbf2-290a2-fda2a8c0@192.168.162.253
User-Agent: Cisco-CUCM8.6
CSeq: 101 OPTIONS
Contact: <192.168.162.253:5060>
Max-Forwards: 0
Content-Length: 0


009884: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
009885: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x2D13478) with key=[918] to table
009886: Mar 26 17:33:38.910: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
009887: Mar 26 17:33:38.910: //-1/000000000000/SIP/Info/ccsip_iwf_init:
009888: Mar 26 17:33:38.910: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
009889: Mar 26 17:33:38.910: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
009890: Mar 26 17:33:38.910: //-1/000000000000/SIP/Info/ccsip_ipip_media_forking_init: MF: Queue is initialised..
009891: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.162.253,Port 42653, Transport 2, SentBy Port 5060
009892: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009893: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009894: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.116.143.151
009895: Mar 26 17:33:38.910: //-1/21203DC48485/SIP/State/sipSPIChangeState: 0x2D13478 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
009896: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.162.253,Port 42653, Transport 2, SentBy Port 5060
009897: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
009898: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.162.253,Port 42653, Transport 2, SentBy Port 5060
009899: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetViaHostInURLFormat: VIA URL Conversion failed for
009900: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: :
009901: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Error/sipSPIGetPeerByCalledPartyId:
input arg error
009902: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
009903: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Error/sipSPIUpdateCallInfo:
input argument error
009904: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
009905: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
009906: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
009907: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
009908: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
009909: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
009910: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
009911: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
009912: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
009913: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 9
009914: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
009915: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
009916: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
009917: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 255
009918: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
009919: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0
009920: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Error/ccsip_ipip_media_forking_anchor_leg_config:
MF: Dial-peer is absent..
009921: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIMFChangeState: MF: Prev state = 0 & New state = -1
009922: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/ccsip_ipip_media_forking_anchor_leg_reset: MF: Anchor leg config reset done...
009923: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: Non dial peer leg - using RTP Supported Codecs
009924: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 18
009925: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 0
009926: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 8
009927: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 9
009928: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 4
009929: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 2
009930: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 15
009931: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/populate_vcc_data: RTP Preferred Codecs supported by GW 255
009932: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
009933: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
009934: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
009935: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled
009936: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
009937: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009938: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009939: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: return addr 10.116.143.151
009940: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.116.143.151
009941: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009942: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009943: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.116.143.151
009944: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_OPTIONS_RESP
009945: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sact_idle_new_message_options: ccsip_api_options_ind returned: SIP_SUCCESS
009946: Mar 26 17:33:38.910: //918/21203DC48485/SIP/State/sipSPIChangeState: 0x2D13478 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OPTIONS_WAIT, SUBSTATE_NONE)
009947: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
009948: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x2D13478 key=49c70a80-1511dbf2-290a2-fda2a8c0@192.168.162.253
009949: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 396 to table
009950: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 38
009951: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPISendOptionsResponse: Associated container=0x13D05438 to Options Response
009952: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Transport/sipSPITransportSendMessage: msg=0xA004028, addr=192.168.162.253, port=42653, sentBy
CDC-CUBE#_port=5060, local_addr=, is_req=0, transport=2, switch=0, callBack=0x88EBB00
009953: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
009954: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
009955: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0xA004028, addr=192.168.162.253, port=42653, local_addr=, connId=3 for TCP
009956: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 396
009957: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[918] removed.
009958: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.
009959: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x2D13478 key=49c70a80-1511dbf2-290a2-fda2a8c0@192.168.162.253
009960: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.
009961: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
009962: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerUnregisterCtxtInConnection: gConnTab=0x264956C, addr=192.168.162.253, port=42653, local_addr=, unregistering context=0x2D13478
009963: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Transport/sipSPITransportContextCleanup: Could not purge context gcb=0x2D13478 from the connection; gcb might be locked
009964: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
009965: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/ccsip_offer_ans_delete:
009966: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/ccsip_iwf_delete:
009967: Mar 26 17:33:38.910: //918/21203DC48485/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 0x2D13478
009968: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[918]
009969: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPITransportSendMessage: gcb has cleanedup, Done with send operations for this context
009970: Mar 26 17:33:38.910: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[918]
009971: Mar 26 17:33:38.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.162.253:5060;branch=z9hG4bK30e8c572db0bb
From: <192.168.162.253>;tag=1151533917
To: <10.116.143.151>;tag=44F9588-17D0
Date: Tue, 26 Mar 2013 17:33:38 GMT
Call-ID: 49c70a80-1511dbf2-290a2-fda2a8c0@192.168.162.253
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 379

v=0
o=CiscoSystemsSIP-GW-UserAgent 3336 3364 IN IP4 10.116.143.151
s=SIP Call
c=IN IP4 10.116.143.151
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.116.143.151
m=image 0 udptl t38
c=IN IP4 10.116.143.151
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

009972: Mar 26 17:33:40.502: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
009973: Mar 26 17:33:40.502: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
009974: Mar 26 17:33:40.502: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x264956C, addr=10.102.243.13, port=39887, local_addr=, connid=4, transport=TCP
009975: Mar 26 17:33:40.502: //-1/xxxxxxxxxxxx/SIP/Info/sipHolderGetConnInstance: Return existing connection for port 39887 connId 4
009976: Mar 26 17:33:40.502: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.116.143.151:5060 SIP/2.0
Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bK7b11fe1da144
From: <10.102.243.13>;tag=1402244681
To: <10.116.143.151>
Date: Tue, 26 Mar 2013 17:33:40 GMT
Call-ID: 4af83780-1511dbf4-4e782-df3660a@10.102.243.13
User-Agent: Cisco-CUCM8.6
CSeq: 101 OPTIONS
Contact: <10.102.243.13:5060>
Max-Forwards: 0
Content-Length: 0


009977: Mar 26 17:33:40.502: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
009978: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x2D13478) with key=[919] to table
009979: Mar 26 17:33:40.506: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
009980: Mar 26 17:33:40.506: //-1/000000000000/SIP/Info/ccsip_iwf_init:
009981: Mar 26 17:33:40.506: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
009982: Mar 26 17:33:40.506: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
009983: Mar 26 17:33:40.506: //-1/000000000000/SIP/Info/ccsip_ipip_media_forking_init: MF: Queue is initialised..
009984: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.102.243.13,Port 39887, Transport 2, SentBy Port 5060
009985: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
009986: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.116.143.151 for SIP
009987: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.116.143.151
009988: Mar 26 17:33:40.506: //-1/2213CC9E8486/SIP/State/sipSPIChangeState: 0x2D13478 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
009989: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.102.243.13,Port 39887, Transport 2, SentBy Port 5060
009990: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
009991: Mar 26 17:33:40.506: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.102.243.13,Port 39887, Transport 2, SentBy Port 5060
009992: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetViaHostInURLFormat: VIA URL Conversion failed for
009993: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: :
009994: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Error/sipSPIGetPeerByCalledPartyId:
input arg error
009995: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
009996: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Error/sipSPIUpdateCallInfo:
input argument error
009997: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
009998: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
009999: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
010000: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
010001: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
010002: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
010003: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
010004: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
010005: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
010006: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 9
010007: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
010008: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
010009: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
010010: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 255
010011: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
010012: Mar 26 17:33:40.506: //919/2213CC9E8486/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0y,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 379

v=0
o=CiscoSystemsSIP-GW-UserAgent 2075 6186 IN IP4 10.116.143.151
s=SIP Call
c=IN IP4 10.116.143.151
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.116.143.151
m=image 0 udptl t38
c=IN IP4 10.116.143.151
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

CDC-CUBE#
010065: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [10.128.0.1]:5060, local_address:[ - ]
010066: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
010067: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
010068: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:02476012665@10.60.34.98;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <02476012665>
From: <>;tag=3573308028-407090
P-Asserted-Identity: <>
Call-ID: 4256141-3573308028-407083@MSX10.gammatelecom.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 10.128.0.1:5060;branch=z9hG4bK9e41074c6ec372ee99914d197f29bc0f
Contact: <>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 195

v=0
o=MSX10 35568351871271985 1 IN IP4 10.128.0.1
s=sip call
c=IN IP4 10.128.0.2
t=0 0
m=audio 10490 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

010069: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
010070: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x2D13478) with key=[920] to table
010071: Mar 26 17:33:48.407: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
010072: Mar 26 17:33:48.407: //-1/000000000000/SIP/Info/ccsip_iwf_init:
010073: Mar 26 17:33:48.407: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
010074: Mar 26 17:33:48.407: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
010075: Mar 26 17:33:48.407: //-1/000000000000/SIP/Info/ccsip_ipip_media_forking_init: MF: Queue is initialised..
010076: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.128.0.1,Port 5060, Transport 1, SentBy Port 5060
010077: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
010078: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.60.34.98 for SIP
010079: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.60.34.98
010080: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/State/sipSPIChangeState: 0x2D13478 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
010081: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.128.0.1,Port 5060, Transport 1, SentBy Port 5060
010082: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
010083: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.128.0.1,Port 5060, Transport 1, SentBy Port 5060
010084: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckIpip: VOIP dialpeer (peer=0x118C5234) found for sip_user: 02476012665
010085: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPICheckAssertedIdConfig: Dialpeer match is not yet done
010086: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIUpdateIdentity: MF: PAI url is present & make a copy..
010087: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIUpdateIdentity: PAI  genUrl is prresent..
010088: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIUpdateIdentity: SIP->PSTN: Setting Calling Number to +447967999481
010089: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIUpdateIdentity: PAI host = 10.128.0.1
010090: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPICheckAssertedIdConfig: Dialpeer match is not yet done
010091: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
010092: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCSTA: No CSTA found in inbound container
010093: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
010094: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x2D13478 key=4256141-3573308028-407083@MSX10.gammatelecom.com02476012665
010095: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
010096: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 02476012665
010097: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: +447967999481
010098: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name , number +447967999481, Calling oct3 0x00, oct_3a 0x81, Called number 02476012665
010099: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetViaHostInURLFormat: VIA URL:sip:10.128.0.1:5060, Host:10.128.0.1
010100: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : +447967999481
010101: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetFromCalledPartyId: P-Called-Party-ID header not found
010102: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetPeerByCalledPartyId: P-Called-Party-ID not found or parse error
010103: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
010104: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Peer tag 101 matched for incoming call
010105: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
010106: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
010107: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address 10.60.34.98 for SIP
010108: Mar 26 17:33:48.407: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 10.60.34.98
010109: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
010110: Mar 26 17:33:48.407: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
010111: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
010112: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
010113: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
010114: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F
010115: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
010116: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0
010117: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
010118: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Incoming: No defer BYE for last
                              call stats
010119: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Media forking disabled
010120: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/ccsip_ipip_media_forking_anchor_leg_config: MF: en_p->encap_s.voIP.voipPeerCfgMediaClass = 0
010121: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/ccsip_ipip_media_forking_anchor_leg_config: MF: Dial-peer has no media class recorder.
010122: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIMFChangeState: MF: Prev state = 0 & New state = -1
010123: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/ccsip_ipip_media_forking_anchor_leg_reset: MF: Anchor leg config reset done...
010124: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/ccsipInitDSCPPolicyInfo: No DSCP Profile configured, No RPH 2 DSCP Mapping and DSCP policing
010125: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIGetCallConfig: Initilise the DSCP policy
010126: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
010127: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
010128: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
010129: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPI_ipip_build_consolidated_header_list: Both passthru and copylist are disabled
010130: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIContinueNewMsgInvite: Calling name , number +447967999481, Calling oct3 0x00, oct_3a 0x81, ext_priv 0x00, Called number 02476012665, oct3 0x00
010131: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
010132: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIContinueNewMsgInvite: Requires reliable-provisional support
010133: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIValidateRequestUri: Not Enabled
010134: Mar 26 17:33:48.411: //-1/26C959BF8487/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
010135: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_IsSDPPassthruEnabled:  - 0
010136: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
010137: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:13D05490
010138: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/Session-Timer/sipSTSLExtractSessionExpiresHdr:
Session-Expires value: 3600 refresher: uac
010139: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/Session-Timer/sipSTSLExtractMinSEHdr: Min-SE Duration: 600
010140: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/Session-Timer/sipSTSLGetInternalSREvent: E_STSL_INITIAL_SR_REQ
010141: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/Session-Timer/sipSTSLInitialSRReqPeerEventGen: sending received session expires to the peer leg
010142: Mar 26 17:33
CDC-CUBE#:48.411: //920/26C959BF8487/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:3600, SE Refresher:uac, Min-SE Value:1800, flags:2001
010143: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/Session-Timer/sipSTSLMain:
        SE: 3600;refresher:uac peer refresher:none, flags:2001, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
        Configured SE:1800, Configured Min-SE:1800
010144: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
010145: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIProcessDiversionHeader: No diversion headers recvd from app container
010146: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIProcessReplacesHeader: No replaces hdr found
010147: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
010148: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIValidateConnectionAddress: Dest port = 10490
SIP: (920) Attribute mid, level 1 instance 1 not found.
010149: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
010150: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling reg_invoke_ip_first_hop()
010151: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: ip_best_local_address 10.60.34.98 for SIP
010152: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: return addr 10.60.34.98
010153: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.60.34.98
010154: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec No Codec
010155: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 96
010156: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved
                          as its in use by other codec No Codec
010157: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (101) is  reserved by another application
010158: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 98
010159: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 101
010160: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 98 for Codec:
010161: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1
010162: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoPtimeNegotiation: No ptime present or multiple ptime attributes that can't be handled
010163: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPISetFaxFlags: FAX_PASSTHROUGH = 0, END_FAX_PASSTHROUGH = 0
010164: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
010165: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec
010166: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec
010167: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 98
010168: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIReserveRtpNtePayload: Reserved the payload type 101 for RTP-NTE
010169: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
010170: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of events.
010171: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: V150 NSE payload = 0, SSE payload = 0, SPRT payload=0
010172: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
010173: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
010174: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
        payload_type=8, codec_bytes=160, codec=g711alaw, dtmf_relay=rtp-nte
        stream_type=voice+dtmf (1), dest_ip_address=10.128.0.2, dest_port=10490
010175: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/State/sipSPIChangeStreamState: Stream (callid =  -1)  State changed from (STREAM_DEAD) to (STREAM_ADDING)
010176: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Media/sipSPIUpdCallWithSdpInfo:
        Preferred Codec        : g711alaw, bytes :160
        Preferred  DTMF relay  : rtp-nte
        Preferred NTE payload  : 101
        Early Media            : No
        Delayed Media          : No
        Bridge Done            : No
        New Media              : No
        DSP DNLD Reqd          : No

010177: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIValidateStreamAddrType: stream:1, Mode : 1
010178: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
010179: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.60.34.98
010180: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/ccsip_offer_ans_handle_rcvd_sdp:
010181: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/ccsip_offer_ans_process_event:
010182: Mar 26 17:33:48.411: //920/26C959BF8487/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_RCVD
010183: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/ccsip_offer_ans_is_invite_offer_valid:  - 1
010184: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/ccsip_offer_ans_common_offer_rcvd_hdlr:
010185: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/ccsip_iwf_handle_network_event:
010186: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/ccsip_iwf_process_event:
010187: Mar 26 17:33:48.411: //920/26C959BF8487/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_RCVD_SDP
010188: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_def_ed_rcvd_sdp_hdlr:
010189: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo:
CallID 920, sdp 0xB44B31C channels 0x2D14BD4
010190: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/copy_channels:
callId 920 size 0 ptr 0x13CDEB70)
010191: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_copy_and_init_channelInfo_params:
CCB t38 version 0 ipip_caps version 0
010192: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_copy_and_init_channelInfo_params:
CCB fax rate 2 ipip_caps rate 14400
010193: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: stream_callid = 920, peer_stream_callid = -1
010194: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo:
Hndl ptype 8 mline 1
010195: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: Selecting codec g711alaw
010196: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/codec_found:
Codec to be matched: 6
010197: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/codec_found:  codecs[i] = 6 & codec = 6 are same..
010198: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 6

010199: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_update_codec_params_in_channelInfo:
010200: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_update_codec_params_in_channelInfo: Parsing from stream media address
010201: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
010202: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_update_codec_params_in_channelInfo:
Adding negotiated codec 6 ptype 8 time 20,   bytes 160 as channel 0 mline 1 ss 1 10.128.0.2:10490
010203: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry
010204: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=0,stream->negotiated_codec_bytes=160,stream->mline_index=1, media_ndx=1
010205: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacCalcAudioBandwidth: calculated audio bandwidth : codec g711alaw      bw 64000
010206: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo:
Hndl ptype 101 mline 1
010207: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_update_dynamic_codec_params:
010208: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_upd_2833_dtmf_params:
010209: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_upd_2833_dtmf_params: setting ipip_caps DTMF to RFC2833: callid = 920, dtmf = 6
010210: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
010211: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry
010212: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[0].codec = 6
010213: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[1].codec = -1
010214: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: srcChannelID = -1, dstChannelID = 920
010215: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice+dtmf
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 920
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 8
          Negotiated Codec       : g711alaw, bytes :160
          Nego. Codec payload    : 8 (tx), 8 (rx)
          Negotiated DTMF relay  : rtp-nte
          Negotiated NTE payload : 101 (tx), 101 (rx)
          Negotiated CN payl
CDC-CUBE#oad  : 0
          Media Srce Addr/Port   : [10.60.34.98]:0
          Media Dest Addr/Port   : [10.128.0.2]:10490

010216: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPI_ipip_calculate_channel_count: Number of streams to report = 1
010217: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count:
Unable to set CHANNEL_COUNT for callid 920
010218: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo:
Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
010219: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: sdp->chnl audio bw 80000 bps video bw 0 bps fax bw 0 bps
010220: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sip_iwf_def_ed_rcvd_sdp_hdlr: Setting media_direction to 0
010221: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacUpdateAccountedBw: bwcac update accounted BW Option 0 flow mode flow-through
audio bw 80000 bps video bw 0 bps fax bw 0 bps total bw 80000 bps  accounted bw 0 bps
010222: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacUpdateInterfaceBw: bwcac acquiring interface GigabitEthernet0/0 bw 80
010223: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacUpdateAccountedBw: bwcac update accounted bw (initial offer) accounted bw set to 80000
010224: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/State/ccsip_cnfsm_debugs: IWF:next_state:S_SIP_IWF_SDP_RCVD_AWAIT_PEER_EVENT
010225: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/State/ccsip_cnfsm_debugs: OA:next_state:S_SIP_EARLY_DIALOG_OFFER_RCVD
010226: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice+dtmf
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 920
          Negotiated Codec       : g711alaw, bytes :160
          Nego. Codec payload    : 8 (tx), 8 (rx)
          Negotiated DTMF relay  : rtp-nte
          Negotiated NTE payload : 101 (tx), 101 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [10.60.34.98]:0
          Media Dest Addr/Port   : [10.128.0.2]:10490

010227: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec       : g711alaw, bytes :160
Preferred Codec        : g711alaw, bytes :160
Preferred  DTMF relay 1 : 6
Preferred  DTMF relay 2 : 0
Negotiated DTMF relay   : 6
Preferred and Negotiated NTE payloads: 101 101
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated V150.1 Modem Passthrough: 0 0
Preferred and Negotiated V150.1 Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0

010228: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacIsDialPeerBwAvailable: bwcac NOP dial-peer bw available tag 101
010229: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacIsInterfaceBwAvailable: bwcac interface bw threshold not configured
010230: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIBwCacVerifyBwThreshold: bwcac verify bw threshold, bw available allow call total bw 80000 bps
010231: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: QOS negotiation for mline_index 1
010232: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIDoStreamQoSNegotiation: Best effort
010233: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
010234: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16478 for stream 1
010235: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=16478
010236: Mar 26 17:33:48.411: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
010237: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 16478
010238: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
SIP update src sdp, negoitated codec 6, payload type 8

010239: Mar 26 17:33:48.411: //920/26C959BF8487/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101

010240: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIUpdateSrcSdpVariablePart: max_event 16
010241: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = 4256141-3573308028-407083@MSX10.gammatelecom.com
010242: Mar 26 17:33:48.415: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCPA: No CPA found in inbound container
010243: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIProcessCPA: No x-cisco-cpa content found
010244: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/ccsip_api_call_setup_ind: Headers from INVITE added to callInfo container
010245: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
010246: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_ExtractPassthruCopyListHdrsFromSipContainer: Unsupported header passthru is not configured and no headers are present in hdr_hash_queue
010247: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_IsContentPassthruEnabled:  - 0
010248: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_ExtractPassthruContentFromSipContainer: Passthru Content Not Enabled
010249: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/ccsip_api_call_setup_ind: Unable to add unsupp headers to container
010250: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/ccsip_api_call_setup_ind: Set Protocol information
010251: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/ccsip_ipip_media_service_get_event_data: Event id = 26
010252: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_antiTrombone: Entered Antitrombone service
010253: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/ccsip_api_call_setup_ind: Setting media_direction to 0
010254: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
010255: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 6,
010256: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_store_config_info: Setting mid_call_config_info = 0x0 for callid = 920
010257: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice+dtmf
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 920
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 8
          Negotiated Codec       : g711alaw, bytes :160
          Nego. Codec payload    : 8 (tx), 8 (rx)
          Negotiated DTMF relay  : rtp-nte
          Negotiated NTE payload : 101 (tx), 101 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [10.60.34.98]:16478
          Media Dest Addr/Port   : [10.128.0.2]:10490

010258: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_calculate_channel_count: Number of streams to report = 1
010259: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPI_ipip_store_channel_info: Setting CHANNEL_COUNT = 1 for callid 920
010260: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIShrlCall: Check peer: 101 for Shared-Line call, callid: 920
010261: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/ccsip_set_bearer_capability:
   Bearer Capability: Speech (0x00)
010262: Mar 26 17:33:48.415: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
010263: Mar 26 17:33:48.415: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
010264: Mar 26 17:33:48.415: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message

010265: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
010266: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 398 to table
010267: Mar 26 17:33:48.415: //920/26C959BF8487/SIP/Info/sipSPIUpdateCallEntry:
Call 920 set InfoType to SPEECH
011227: Mar 26 17:33:48.439: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstancePurgeContextInConnection: Purging context gcb=0x2D13478 from the connection=0xAFF7CE4 context list
011228: Mar 26 17:33:48.439: //920/26C959BF8487/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
011229: Mar 26 17:33:48.439: //920/26C959BF8487/SIP/Info/ccsip_offer_ans_delete:
011230: Mar 26 17:33:48.439: //920/26C959BF8487/SIP/Info/ccsip_iwf_delete:
011231: Mar 26 17:33:48.439: //920/26C959BF8487/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 0x2D13478
011232: Mar 26 17:33:48.439: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[920]
CDC-CUBE#term no mon
CDC-CUBE#

khayes1984
Level 1
Level 1

I'm dealing with something similar to this but here's my flow

ISP/ITSP->ASA-CUBE->CUCMBE

Outbound calls work fine, it's inbound that's getting me...here's my config.

voice translation-rule 10

rule 1 /^\+1\(..........\)$/ /\1/

!

!

voice translation-profile in_rmv_plus1

translate called 10

!

!
dial-peer voice 100 voip
description Incoming dialpeer and 1+10 digits to Intelepeer
translation-profile incoming in_rmv_plus1
destination-pattern ^1[2-9]..[2-9]......$
session protocol sipv2
session target sip-server
session transport tcp
incoming called-number .T
voice-class codec 1 
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 101 voip
description 10-digit local calls to Intelepeer
destination-pattern ^[2-9]..[2-9]......$
session protocol sipv2
session target sip-server
session transport tcp
voice-class codec 1 
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 102 voip
description International calls to Intelepeer
destination-pattern ^011T
session protocol sipv2
session target sip-server
session transport tcp
voice-class codec 1 
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 103 voip
description N11 calls to Intelepeer
destination-pattern ^[2-9]11$
session protocol sipv2
session target sip-server
session transport tcp
voice-class codec 1 
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 300 voip
description 20890138XX calls to primary CUCM
huntstop
preference 1
destination-pattern ^20890138..$
session protocol sipv2
session target ipv4:192.168.1.8
voice-class codec 1 
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 302 voip
description Incoming calls from primary CUCM
huntstop
session protocol sipv2
session target ipv4:192.168.1.8
incoming called-number .
voice-class codec 1 
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
!
gateway
timer receive-rtp 1200
!
sip-ua
retry invite 2
retry bye 2
retry cancel 2
retry register 10
sip-server ipv4:68.68.120.82
!

I don't require a UA username and password..

****Debug Output******

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=+12089013875, Called Number=+12089013875, Peer Info Type=DIALPEER_INFO_SPEECH

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=+12089013875

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=+12089013875, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=+12089013875, Called Number=+12089013875, Peer Info Type=DIALPEER_INFO_SPEECH

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=+12089013875

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=+12089013875, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Mar 26 17:12:21.506: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=+12089013875, Called Number=+12089013875, Peer Info Type=DIALPEER_INFO_SPEECH

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=+12089013875

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=+12089013875, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Mar 26 17:12:17.606: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=+12089013875, Called Number=+12089013875, Peer Info Type=DIALPEER_INFO_SPEECH

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=+12089013875

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Mar 26 17:12:21.502: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=+12089013875, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

Mar 26 17:12:21.506: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

Kenneth,

Open another discussion for your issue...So we can look at that seperately

Please rate all useful posts

Hi Aok,

As you said I have attached debug file and please see below dial-peer config

dial-peer voice 101 voip
description *** Inbound Calls from ITSP ***
translation-profile incoming SIP-TRUNK-IN
session protocol sipv2
incoming called-number 02476012665
dtmf-relay rtp-nte
codec g711alaw
!
dial-peer voice 20 voip
description **** Outbound calls to Primary CUCM ***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:10.102.243.13
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 21 voip
description *** Outbound Calls to Backup CUCM***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:10.102.243.12
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 22 voip
description *** Outbound LAN side dial-peer ***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:192.168.132.2
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 23 voip
description *** Outbound LAN side dial-peer ***
destination-pattern 0[2-9].........
session protocol sipv2
session target ipv4:192.168.162.253
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 1 voip
description *** Inbound Calls from ITSP ***
session protocol sipv2
incoming called-number [2-9].........
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 2 voip
description *** Outbound calls to ITSP ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw

Your logs are not complete...

Please configure buffer loggin as follows

logging buffered 10000000

no logging rate-limit

no logging monitor

Then do a test call again and once the call has finished, use software like putty to loggin to the gateway..Configure putty to log files to your desktop or anywhere you can access it and then do the ff:

term len 0

sh loggin

All the logs in the buffer will then be sent to the putty file. Send that file to me

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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Muhhamad,

What ip address is 10.160.134.98 Is this your CUBE IP address? I thought this is the provider's ip address?

The invite is sent to this ip address..Is this your CUBE IP address?

INVITE sip:02476012665@10.60.34.98

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