About Mark

By Mark

Mark Holloway (CCIE #27384)

I started this blog as a place holder for my technical notes.  I had a difficult time finding resources on the Internet for SIP related technologies specific to Service Provider and large Enterprise VoIP deployments. I knew that if I write my notes on paper they would get lost, so I created a WordPress blog for my own personal storage of information. I soon realized this is a superior method to store notes and I would no longer invest in spiral bound notebooks. I started receiving messages from folks who work with similar technologies who found the information I am writing about to be useful for them as well. I continue to write about my technical experiences and that more people will find this information equally as useful for them as it is for me.

 

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  1. Tom says:

    Mark… I have a small company that installs IP-PBX systems and SIP Trunking in hotels in the U.S. nationwide. I am looking for someone to remotely program Adtran 900 series units to act as IP gateways to legacy PBX systems. Do you freelance and would you be interested in assisting us in our SIP Trunking roll-outs.

    Tom

  2. Jeff C says:

    Mark,

    Always have enjoyed the site. Having a problem using max concurrent calls on Acme/Oracle ASC. Would you consider a consulting engagement to assist? May very well take you 20 minutes to remedy, not looking for long term.

    Send me an email if it’s something you’d be interested in.

    In a nutshell, we’re using max-number-of-concurrent-calls in dial-plan/route definition to limit calls. Problem occurs when calls then get referred over to another endpoint. The call is still up but is no longer counted on the incoming route so max call setting becomes useless.

    Thanks,
    Jeff

  3. Allan Mancera says:

    Hey Mark

    Thanks for all the guides you upload, your site is my #1 resource for CUCM and Acme.

    Can you help me on finding a guide on how to use Local Route Tables? I’ve been reading a lot of documentation but there is not one example or anything I could use to do some testing. Right now I’m trying to setup an Acme SBC to route calls between PSTN – CUCM – Avaya SM and I don’t really want to do all the local-policy for each realm and appliance. Could you help me?

    In a second question, I’m getting really interested in certifying in Acme, do you know a good document resource so I can start studying?

    Thank you for all the work and the site.

    Best regards

    Allan Mancera

  4. Vincent says:

    Hi Mark
    We have a call being routed via two SBC’s, both parties are in the same subnet, so I want to release the RTP from the SBC, do you know the SIP Options for the config?

    “Remember that for this feature to work, you must
    also set the options parameter in the SIP configuration accordingly.”
    Config I found. msm-release—Enable or disable the inclusion of multi-system (multiple Net-
    Net SBCs) media release information in the SIP signaling request sent into the
    realm identified by this realm-config element. If this field is set to enabled,
    another Net-Net SBC is allowed to decode the encoded SIP signaling request
    message data sent from a SIP endpoint to another SIP endpoint in the same
    network to resore the original SDP and subsequently allow the media to flow
    directly between those two SIP endpoints in the same network serviced by
    multiple Net-net SBCs. If this field is disabled, the media and signaling will pass
    through boht Net-Net SBCs. Remember that for this feature to work, you must
    also set the options parameter in the SIP configuration accordingly. The default
    is disabled. Valid values are:

  5. Greg Wilson says:

    Mark, Good day! Can you provide guidance on modifying FTP access credentials in a Acme 4500 running 6.4 code? I have a client who wants read-only access to a managed SBC; however, I want to make sure ONLY READ ONLY is permitted ANYWHERE! Please; and, thanks in advance — wilson

  6. Greg Wilson says:

    Mark, question #2? I have a customer network with a lot of fragmented invites. We have fragment-msg-bandwidth set to 20% of total bandwidth in a Acme 4500 / rel 6.4. Is there a way to query the system to see bandwidth allocation; more importantly if we hare hitting that 20%. We are seeing fragmented hold/transfer RE-INVITEs not getting a Acme response. Thank you sir!!

  7. CSG says:

    Mark, I have a large Carrier that has many NN4500 Oracle Acme Packet 4500 with unlimited sessions available for sale. Please pass this on to anyone you feel might have interest. Also they have the RTU transferable by the carrier. NICE SITE!

    Thank you
    Lee

  8. dramhit says:

    Hello Mark,
    Can you please help me to understand what will be the effect of disabling auto-config on oracle SBC.

    Regards
    Damian

  9. Bud says:

    Hi Mark
    I have to move the management interface on an Acme Net-Net 3820 from the voice vlan to the data vlan. could I trouble you with the commands and or process? Is it service affecting (as in a required reboot – etc) ?
    My Google searches keep bringing me to your site so I thought I would ask “the Master”.
    Thanks,
    Bud

  10. JéJé says:

    Hi Mark,
    i have just received 2 Net-Net 3820, and i want to make the first configuration (like i dot on the VM version before).
    First, i “setup product” to “SBC”, then “setup entitlements”. My bootparam is correctly set. I can access to the web GUI, but i have not the menu “configuration”.

    Does i have to activate specific licence on my appliance to have the full web gui ?
    Does my appliance correctly set and fully operaionnal with the two “setup” ?

    I don’t understant everything about licencing …

    Thanks for your help,
    regards.

  11. JéJé says:

    Note: the version i use is the Acme Packet Net-Net 3820 SCZ7.2.0 MR-2 GA (Build 252)

    Boot line: wancom(0,0)acme1:/boot/nnSCZ720m2.32.bz e=192.168.1.1:ffffff00 g=192.168.1.254 u=vxftp pw=vxftp tn=ACME1

  12. David says:

    Hi Mark

    quick question how do we export/ what is the quickest way to export local-policy numbers configured on an SBC to a excel or text file?

    Kind Regards

  13. Jon says:

    Hi, found your blog and I’m totally new to ACME SBC’s. I’m just trying to get the basics, can you recommend any site or books. I already read SBC for dummies, thx!

    Jon

  14. Jon says:

    Forgot to add, this is for ACME SBC in a Avaya Session Manager 6.0 environment.

  15. Sajish says:

    Hey Mark,
    We have the SIP trunk delivered recently its internet based SIP no autherntication, i am trying to make the outbound working for the SIP trunk.. incoming works fine… below is the output of the debug and config.

    calling number – 4580251197
    called number – 353861300479

    dial-peer hunt 0
    AD PRE PASS OUT
    TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE
    2 voip up up 0 syst
    100 voip up up ^…….$ 1 syst ipv4:10.43.32.84
    101 voip up up ^…….$ 2 syst ipv4:10.35.32.85
    103 voip up up ^…….$ 3 syst ipv4:10.43.32.83
    200 voip up up .T 0 syst ipv4:64.215.222.109:
    201 voip up up 353861300479 0 syst ipv4:64.215.222.109:
    SIP-Router-DB#
    SIP-Router-DB#
    SIP-Router-DB#
    SIP-Router-DB#ter
    SIP-Router-DB#terminal moni
    SIP-Router-DB#terminal monitor
    SIP-Router-DB#
    *Aug 2 09:15:01.722: //-1/F4CB47000000/CCAPI/cc_api_display_ie_subfields:
    cc_api_call_setup_ind_common:
    cisco-username=4580251197
    —– ccCallInfo IE subfields —–
    cisco-ani=sip:4580251197@10.43.32.84
    cisco-anitype=0
    cisco-aniplan=0
    cisco-anipi=0
    cisco-anisi=1
    dest=sip:353861300479@10.35.61.15:5060
    cisco-desttype=0
    cisco-destplan=0
    cisco-rdie=FFFFFFFF
    cisco-rdn=
    cisco-rdntype=0
    cisco-rdnplan=0
    cisco-rdnpi=-1
    cisco-rdnsi=-1
    cisco-redirectreason=-1 fwd_final_type =0
    final_redirectNumber =
    hunt_group_timeout =0

    *Aug 2 09:15:01.722: //-1/F4CB47000000/CCAPI/cc_api_call_setup_ind_common:
    Interface=0x49EE124C, Call Info(
    Calling Number=sip:4580251197@10.43.32.84,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=sip:353861300479@10.35.61.15:5060(TON=Unknown, NPI=Unknown),
    Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
    Incoming Dial-peer=2, Progress Indication=NULL(0), Calling IE Present=TRUE,
    Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=1074
    *Aug 2 09:15:01.722: //-1/F4CB47000000/CCAPI/ccCheckClipClir:
    In: Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    *Aug 2 09:15:01.722: //-1/F4CB47000000/CCAPI/ccCheckClipClir:
    Out: Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    *Aug 2 09:15:01.722: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:01.726: :cc_get_feature_vsa malloc success
    *Aug 2 09:15:01.726: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:01.726: cc_get_feature_vsa count is 1
    *Aug 2 09:15:01.726: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:01.726: :FEATURE_VSA attributes are: feature_name:0,feature_time:1270430568,feature_id:519
    *Aug 2 09:15:01.726: //1074/F4CB47000000/CCAPI/cc_api_call_setup_ind_common:
    Set Up Event Sent;
    Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=(TON=Unknown, NPI=Unknown))
    *Aug 2 09:15:01.726: //1074/F4CB47000000/CCAPI/cc_process_call_setup_ind:
    Event=0x4B3EC378
    *Aug 2 09:15:01.726: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
    Try with the demoted called number 353861300479
    *Aug 2 09:15:01.730: //1074/F4CB47000000/CCAPI/ccCallSetContext:
    Context=0x4BBBE760
    *Aug 2 09:15:01.730: //1074/F4CB47000000/CCAPI/cc_process_call_setup_ind:
    >>>>CCAPI handed cid 1074 with tag 2 to app “_ManagedAppProcess_Default”
    *Aug 2 09:15:01.730: //1074/F4CB47000000/CCAPI/ccCallProceeding:
    Progress Indication=NULL(0)
    *Aug 2 09:15:01.730: //-1/xxxxxxxxxxxx/CCAPI/ccGetMemPoolFromContainer:
    mempool not found from usrContainer(4B2F25C4)
    *Aug 2 09:15:01.730: //-1/xxxxxxxxxxxx/CCAPI/ccCreateMemPoolInContainer:
    Mempool(4B37E718) created in usrContainer(4B2F25C4)
    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/ccCallSetupRequest:
    Destination=, Calling IE Present=TRUE, Mode=0,
    Outgoing Dial-peer=201, Params=0x4BBAB070, Progress Indication=NULL(0)
    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/ccCheckClipClir:
    In: Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/ccCheckClipClir:
    Out: Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/ccCallSetupRequest:
    Destination Pattern=353861300479, Called Number=353861300479, Digit Strip=FALSE
    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/ccCallSetupRequest:
    Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=353861300479(TON=Unknown, NPI=Unknown),
    Redirect Number=, Display Info=Sajish Denmark
    Account Number=4580251197, Final Destination Flag=TRUE,
    Guid=F4CB4700-0001-0000-0000-003254202B0A, Outgoing Dial-peer=201
    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/cc_api_display_ie_subfields:
    ccCallSetupRequest:
    cisco-username=4580251197
    —– ccCallInfo IE subfields —–
    cisco-ani=sip:4580251197@10.43.32.84
    cisco-anitype=0
    cisco-aniplan=0
    cisco-anipi=0
    cisco-anisi=1
    dest=353861300479
    cisco-desttype=0
    cisco-destplan=0
    cisco-rdie=FFFFFFFF
    cisco-rdn=
    cisco-rdntype=0
    cisco-rdnplan=0
    cisco-rdnpi=-1
    cisco-rdnsi=-1
    cisco-redirectreason=-1 fwd_final_type =0
    final_redirectNumber =
    hunt_group_timeout =0

    *Aug 2 09:15:01.734: //1074/F4CB47000000/CCAPI/ccIFCallSetupRequestPrivate:
    Interface=0x49DEECBC, Interface Type=1, Destination=, Mode=0x0,
    Call Params(Calling Number=sip:4580251197@10.43.32.84,(Calling Name=Sajish Denmark)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=353861300479(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
    Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=201, Call Count On=FALSE,
    Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
    *Aug 2 09:15:01.734: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:01.734: :cc_get_feature_vsa malloc success
    *Aug 2 09:15:01.734: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:01.734: cc_get_feature_vsa count is 2
    *Aug 2 09:15:01.734: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:01.734: :FEATURE_VSA attributes are: feature_name:0,feature_time:1270431016,feature_id:520
    *Aug 2 09:15:01.734: //1075/F4CB47000000/CCAPI/ccIFCallSetupRequestPrivate:
    SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
    *Aug 2 09:15:01.738: //1075/F4CB47000000/CCAPI/ccCallSetContext:
    Context=0x4BBAB020
    *Aug 2 09:15:01.738: //1074/F4CB47000000/CCAPI/ccSaveDialpeerTag:
    Outgoing Dial-peer=201
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/cc_api_call_disconnected:
    Cause Value=38, Interface=0x49DEECBC, Call Id=1075
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/cc_api_call_disconnected:
    Call Entry(Responsed=TRUE, Cause Value=38, Retry Count=0)
    *Aug 2 09:15:16.738: //1074/F4CB47000000/CCAPI/ccCallReleaseResources:
    release reserved xcoding resource.
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/ccCallSetAAA_Accounting:
    Accounting=1, Call Id=1075
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/ccCallDisconnect:
    Cause Value=38, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=38)
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/ccCallDisconnect:
    Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38)
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/cc_api_get_transfer_info:
    Transfer Number Is Null
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/cc_api_call_disconnect_done:
    Disposition=0, Interface=0x49DEECBC, Tag=0x0, Call Id=1075,
    Call Entry(Disconnect Cause=38, Voice Class Cause Code=0, Retry Count=0)
    *Aug 2 09:15:16.738: //1075/F4CB47000000/CCAPI/cc_api_call_disconnect_done:
    Call Disconnect Event Sent
    *Aug 2 09:15:16.738: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

    *Aug 2 09:15:16.738: :cc_free_feature_vsa freeing 4BB93D20
    *Aug 2 09:15:16.738: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

    *Aug 2 09:15:16.738: vsacount in free is 1
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccCallSetupRequest:
    Destination=, Calling IE Present=TRUE, Mode=0,
    Outgoing Dial-peer=200, Params=0x4BBC3AB0, Progress Indication=NULL(0)
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccCheckClipClir:
    In: Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccCheckClipClir:
    Out: Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccCallSetupRequest:
    Destination Pattern=.T, Called Number=353861300479, Digit Strip=FALSE
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccCallSetupRequest:
    Calling Number=sip:4580251197@10.43.32.84(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=353861300479(TON=Unknown, NPI=Unknown),
    Redirect Number=, Display Info=Sajish Denmark
    Account Number=4580251197, Final Destination Flag=TRUE,
    Guid=F4CB4700-0001-0000-0000-003254202B0A, Outgoing Dial-peer=200
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/cc_api_display_ie_subfields:
    ccCallSetupRequest:
    cisco-username=4580251197
    —– ccCallInfo IE subfields —–
    cisco-ani=sip:4580251197@10.43.32.84
    cisco-anitype=0
    cisco-aniplan=0
    cisco-anipi=0
    cisco-anisi=1
    dest=353861300479
    cisco-desttype=0
    cisco-destplan=0
    cisco-rdie=FFFFFFFF
    cisco-rdn=
    cisco-rdntype=0
    cisco-rdnplan=0
    cisco-rdnpi=-1
    cisco-rdnsi=-1
    cisco-redirectreason=-1 fwd_final_type =0
    final_redirectNumber =
    hunt_group_timeout =0

    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccIFCallSetupRequestPrivate:
    Interface=0x49EE124C, Interface Type=3, Destination=, Mode=0x0,
    Call Params(Calling Number=sip:4580251197@10.43.32.84,(Calling Name=Sajish Denmark)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
    Called Number=353861300479(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
    Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=200, Call Count On=FALSE,
    Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
    *Aug 2 09:15:16.742: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:16.742: :cc_get_feature_vsa malloc success
    *Aug 2 09:15:16.742: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:16.742: cc_get_feature_vsa count is 2
    *Aug 2 09:15:16.742: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

    *Aug 2 09:15:16.742: :FEATURE_VSA attributes are: feature_name:0,feature_time:1270431016,feature_id:521
    *Aug 2 09:15:16.742: //1076/F4CB47000000/CCAPI/ccIFCallSetupRequestPrivate:
    SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
    *Aug 2 09:15:16.742: //1076/F4CB47000000/CCAPI/ccCallSetContext:
    Context=0x4BBC3A60
    *Aug 2 09:15:16.742: //1074/F4CB47000000/CCAPI/ccSaveDialpeerTag:
    Outgoing Dial-peer=200
    *Aug 2 09:15:16.746: //1076/F4CB47000000/CCAPI/cc_api_call_proceeding:
    Interface=0x49EE124C, Progress Indication=NULL(0)
    *Aug 2 09:15:16.890: //1076/F4CB47000000/CCAPI/cc_api_call_disconnected:
    Cause Value=1, Interface=0x49EE124C, Call Id=1076
    *Aug 2 09:15:16.890: //1076/F4CB47000000/CCAPI/cc_api_call_disconnected:
    Call Entry(Responsed=TRUE, Cause Value=1, Retry Count=0)
    *Aug 2 09:15:16.894: //1074/F4CB47000000/CCAPI/ccCallReleaseResources:
    release reserved xcoding resource.
    *Aug 2 09:15:16.894: //1076/F4CB47000000/CCAPI/ccCallSetAAA_Accounting:
    Accounting=0, Call Id=1076
    *Aug 2 09:15:16.894: //1076/F4CB47000000/CCAPI/ccCallDisconnect:
    Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=1)
    *Aug 2 09:15:16.894: //1076/F4CB47000000/CCAPI/ccCallDisconnect:
    Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
    *Aug 2 09:15:16.894: //1076/F4CB47000000/CCAPI/cc_api_call_disconnect_done:
    Disposition=0, Interface=0x49EE124C, Tag=0x0, Call Id=1076,
    Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
    *Aug 2 09:15:16.894: //1076/F4CB47000000/CCAPI/cc_api_call_disconnect_done:
    Call Disconnect Event Sent
    *Aug 2 09:15:16.894: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

    *Aug 2 09:15:16.894: :cc_free_feature_vsa freeing 4BB93D20
    *Aug 2 09:15:16.894: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

    *Aug 2 09:15:16.894: vsacount in free is 1
    *Aug 2 09:15:16.894: //1076/F4CB47000000/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4B8DBEF0
    State of The Call : STATE_DEAD
    TCP Sockets Used : NO
    Calling Number : 4580251197
    Called Number : 353861300479
    Source IP Address (Sig ): 89.101.141.200
    Destn SIP Req Addr:Port : 64.215.222.109:5060
    Destn SIP Resp Addr:Port : 64.215.222.109:5060
    Destination Name : 64.215.222.109

    *Aug 2 09:15:16.894: //1076/F4CB47000000/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : No Codec
    Negotiated Codec Bytes : 0
    Nego. Codec payload : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0 (tx), 0 (rx)
    Source IP Address (Media): 89.101.141.200
    Source IP Port (Media): 18440
    Destn IP Address (Media): –
    Destn IP Port (Media): 0
    Orig Destn IP Address:Port (Media): [ – ]:0

    *Aug 2 09:15:16.894: //1076/F4CB47000000/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC) : 1
    Disconnect Cause (SIP) : 404

    *Aug 2 09:15:16.898: //1074/F4CB47000000/CCAPI/ccCallDisconnect:
    Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
    *Aug 2 09:15:16.898: //1074/F4CB47000000/CCAPI/ccCallDisconnect:
    Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
    *Aug 2 09:15:16.898: //-1/xxxxxxxxxxxx/CCAPI/ccMemPoolTDFreeHelper:
    data = 4B2FE8F8
    *Aug 2 09:15:16.898: ccMemPoolTDFreeHelper:mem_mgr_mempool_free: mem_refcnt(4B37E718)=0 – mempool cleanup
    *Aug 2 09:15:16.942: //1074/F4CB47000000/CCAPI/cc_api_call_disconnect_done:
    Disposition=0, Interface=0x49EE124C, Tag=0x0, Call Id=1074,
    Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
    *Aug 2 09:15:16.946: //1074/F4CB47000000/CCAPI/cc_api_call_disconnect_done:
    Call Disconnect Event Sent
    *Aug 2 09:15:16.946: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

    *Aug 2 09:15:16.946: :cc_free_feature_vsa freeing 4BB93B60
    *Aug 2 09:15:16.946: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

    *Aug 2 09:15:16.946: vsacount in free is 0
    *Aug 2 09:15:16.946: //1074/F4CB47000000/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4B8D64F8
    State of The Call : STATE_DEAD
    TCP Sockets Used : NO
    Calling Number : 4580251197
    Called Number : 353861300479
    Source IP Address (Sig ): 10.35.61.15
    Destn SIP Req Addr:Port : 10.43.32.84:5060
    Destn SIP Resp Addr:Port : 10.43.32.84:5060
    Destination Name : 10.43.32.84

    *Aug 2 09:15:16.946: //1074/F4CB47000000/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : No Codec
    Negotiated Codec Bytes : 0
    Nego. Codec payload : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0 (tx), 0 (rx)
    Source IP Address (Media): 10.35.61.15
    Source IP Port (Media): 17040
    Destn IP Address (Media): –
    Destn IP Port (Media): 0
    Orig Destn IP Address:Port (Media): [ – ]:0

    *Aug 2 09:15:16.946: //1074/F4CB47000000/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC) : 1
    Disconnect Cause (SIP) : 404

    *Aug 2 09:18:16.946: //-1/xxxxxxxxxxxx/CCAPI/ccAppShutdownMode:
    ccAppShutdownMode: remove it from the queue

  16. Mike Perri says:

    Trying one more time

    Mark,

    Working a sip-manipulation for SBC (ACME) trying to strip digits away on the Refer-To headed and I can not seem to nail it down.

    header-rule
    name referto499
    header-name Refer-To
    action manipulate
    comparison-type pattern-rule
    msg-type any
    methods REFER
    match-value any
    new-value
    element-rule
    name ReqURIstrip499
    parameter-name
    type uri-user
    action replace
    match-val-type any
    comparison-type pattern-rule
    match-value
    new-value $ORIGINAL-^”499″

    I am applying that header to the out-manipulationid on the SA I route back to my customer.

    The Refer-To header still states the following

    ##(( sip.Refer-To == “” ))##

    We need that 499 stripped for the transfer to be completed on this REFER SIP message.

    Any ideas?

  17. Mark says:

    The Acme Packet SBC documentation and integration guides are publicly available on the Oracle web site. Check out the Enterprise Communications section http://docs.oracle.com/en/industries/communications/

  18. Andrey Drofa says:

    A good day Mark, thanks for your work and your blog! I have a problem with my CME Router 2911. I have a topology sip trunk to ISP and another trunk to CallCenter with BGP routing with him. I have a trouble with SIP SDP, when I have a local ip address connection information from CME, following is SDP offer in 200 OK sent by Router/CME. How I can change the local address in SDP?

    With Regards Andrey Drofa
    Saint-Petersburg, Russia

  19. Jeff G says:

    Rats. I hate it when my “click submit” finger is faster than the text editor part of my brain…

    Mark-

    I actually posted that last without thanking you for the resources you have put up. They’re the best I’ve found, by far. They’ve taught a lot about how to do individual things, but I’m pretty sure that I have some big knowledge gaps in between. That’s what I’m looking for… a “dummies” type of guide to tie the pieces together.

    Jeff

  20. Mark says:

    You can download the VM version of the SBC for free from http://edelivery.oracle.com and it includes an OVA template. All the documentation is publicly available here http://docs.oracle.com/en/industries/communications

    If you want to see a step by step breakdown of how to configure an SIP Trunk (AT&T) with CUCM then check out this guide http://www.oracle.com/webfolder/technetwork/acmepacket/Cisco/att-ipflex-cisco-ucm6-ap-3k-4k-sbc.pdf

  21. Non tar diags for 4500 says:

    Scott,
    I’m trying to help reset some passwords on some used-market 4500s that have an old bootloader that won’t read tar versions of diags. Do you have an older version? Oracle support doesn’t have any in their patch inventory and am trying to NOT open a case on units not under support.
    Thx

  22. Anthony says:

    Hi Mark,

    Trying to understand local policies and order of president. Acme training taught me one thing but a tech I just talked to at Oracle and he said something different. I’m using local policies to route outbound traffic to different carriers based on destination area code and sometimes they overlap. My understanding was it would first match on the to-address field and if there was more than one match, it would pick the more specify one. The tech at Oracle said it wouldn’t pick the more specific / better match and instead all matching local policies are considered and the next thing it looks at is the cost in policy-attributes. Can you tell me what / who is correct please.
    Thanks!

  23. Ralph Hope says:

    The SBC is being presented with an INVITE that has “sips” as the scheme in the request-uri:

    sips:2025551234@:5061;transport=tls SIP/2.0

    This invite is routed via local-policy to another interface to a VoIP switch that will not accept sips, so I am trying to figure out how to manipulate the “sips” to “sip” in sip-manipulation so that the invite request-uri would look like this:

    sip:2025551234@:5061;transport=tls SIP/2.0

    I have tried a number of manipulations in the element-rule for the requri parameter-name, but I have so far been unsuccessful in accomplishing this manipulation to sip. I would welcome any advice on how to progress with this.

    Thank you very much for your attention and assistance.

    Ralph Hope

  24. dan says:

    mark quick question in relation to a SIP trunk from CUBE to provider.

    I register to provider with 2107214100@provider.voice.net

    however the DID that i’m assigned is 5552221212

    the problem i see is that when i call inbound no matter what inbound DID i dial, it seems that the DID is not delivered and call comes in as 2107214100.

    i think i have to do something with SIP-Profiles or what not and thought you may have a clue to assist. cheers

    dan

  25. Charlie Davenport says:

    Hey Mark,

    Wondering if you are available to email off list? Please let me know.

  26. Luis Veiga says:

    Hello Mark,
    i have a question. Could you please give me some clue?

    I have two SBCs(one 4250 and one 4500) and wondering if is possible to do a transfering of xml config files from 4250 with about 900 sip trunks, to the other 4500 with about 600 sip trunks?

    I’m thinking to exclude all commum elements on 4250 box like systemConfig, authentication, sipManipulation, phyInterface, RedundancyConfig, snmpCommunity and finaly, add to xml 4500 config file all realmConfig, localPolicy, sipInterface, sessionAgent, sessionAgentGroups, networkInterfaces and ipRoute…

    Paste-Config is not an option because my 4250’s ruuning config have more than 8 Mbytes

    Do you see any problem on this procedure? All 4250 sip-trunks wiil work on 4500 box?
    4250 is running 6.2.0 version.
    4500 is running 7.3.0 Linux version.

    I saw that both xml files have the same xml tags. Wondering if objectID will be a problem.

    Thanks for any help or clue to do that

    Best Regards,
    Luis Veiga

  27. Pete says:

    Hello Mark. I have a problem with one of my ACME 4500s that I’m not able to resolve. I have an HA pair running scz6.4 that I’m attempting to upgrade to 7.4 – this only affects one of the units in the pair, the other is just fine. I’m able to upload the new bootloader and code files to flash via ftp but when I use the cli and do a ‘show dir /boot’, etc. I don’t see all of the files – this is the same for the other directories on the unit. I’m able to manipulate the files as needed with WinSCP but nothing I do makes them show up via CLI. Same goes in the shell. A reboot did not resolve this. This is preventing me from upgrading the code as it can’t see the new files.

    I have an SR open wth Oracle but so far nothing – any ideas?

    Thanks in advance,

    Pete

    Network Engineer
    CBTS

  28. Mike says:

    Hey Mark, I was trying to setup a test CUCM at the house to screw around on. Do you know where I can snag a free iso? Thanks!

  29. Blake Willemsen says:

    Hey Mark, awesome info here. Do you do any kind of work on the side. We could use some help with our 3820’s.

  30. Sławek says:

    Hi Mark,

    I hope you are doing well. I read your article about SCCP to SIP reconfiguration
    http://www.markholloway.com/blog/?p=549

    I have some issue during this process. I copied every firmware files into TFTP server directory and phone (Cisco 7965G) is downloading default term65.loads file, then after reboot it stays on this firmware- it didn’t download software what I wrote down in XMLdefault file (SIP firmware).

    Do you know what can be the issue? Will you be able to help me in that case?

    Thank you for your time and I really appreciate any help. I saw that you have your own company. If you would like to do this as a service you can also send me your proposition.

    Best regards,
    Sławek.

  31. Saravana Pandian says:

    Hi Mark,

    Do you have idea for configuring the SIP manipulation for conversing Call info to Diversion.

    Regards,
    Saravana

  32. Brian says:

    Hi Mark,

    Do you have any experience, advice or opinions on “ChatOps” or remote config change on Acme SBCs?

    Use Case – Calls failing over carrier 1, send text message or chat command “move all traffic on SBCa to carrier 2” – remotely execute command that moves traffic on SBCa.

    Thanks in advance!

    -Brian

  33. Gary says:

    Hi Mark a great site , always looked over it to find solutions to problems.
    One I have now is we are upgrading the network, so replacing a 4500 with a 4600.

    So is the any things I need to look out for on the 4600?

    and I need to get the config on the 4500 on the 4600
    can I create a back-up of the config from the 4500, ftp the file to the 4600, and just restore/load?
    Is there a tool that I can run the Config text or back up file through , to generate the 4600 config to load?
    noted the MAC addresses on the interfaces and system-config would need up dating for the 4600

  34. Open question to Mark and whoever wants to contribute.

    I am looking for a ‘clear’ way to document the configuration of the SBCs. Unfortunately this documentation will have to be readable by non SBC savi personnel.

    Does anybody have any ideas of a way to layout a config, as the ‘show run’ is not the clearest ?

    Paul.

  35. Luiz Carlos says:

    How configuration the SBC 6300 for transfer protocol UUI ?

  36. PacketCat says:

    Paul —

    If show run is not the clearest / you are worried about the recipient not being able to infer the necessary information from a config dump, then you’ll need to first determine what the “relevant data” is that your audience cares about, and provide them just that.

    My advice — abstract the specifics of what your intended audience might care about — e.g., the session agents of a given SBC, its functional role in your network, local routing policies, etc — then write a parser that when fed the configuration output, formats relevant data into just the specific things that your audience cares about.

    For example, in my network, SBCs may fall into one of several templated roles, such as access/registration proxy, or peering with external systems — I might want to be able to provide an interface or documentation of a list of of the peers on that SBC — so my parser script would identify all of the SBCs that fill that peering role, take a config output of their session agents, capture just the relevant information (session agent name, IP address (or list of NAPTR resolved agents if applicable), and anything else my audience might care about), then format it in some manner that will be meaningful to whoever you’re presenting to. If it’s a developer somewhere I’ll feed them JSON, if it’s a manager, I might use some formatting library to make it prettier, etc ..