RV016 incomming sip-applications

Nice day

I have a bad router Cisco RV016 framework to properly manage SIP applications.

Here what I have:

LAN - 10.1.0.0/24

10.1.0.1 - IP LAN of the RV016

10.1.0.2 - IP on the NETWORK adapter for the software IP - PBX "3CX Phone SYSTEM"

RV016 have two interfaces configured, WAN to work with various Internet service providers.

I also have a VoIP service provider somewhere through the internet, which is handlling the incoming and outgoing calls.

For incomming calls VoIP to work properly, I did some port-forwarding rules:

SIP - all connections inbound interface WAN and destination port 5060 are translated to the 10.1.0.2

RTP - incommming connections WAN interface and the destination ports 9000-9015 are translated to the 10.1.0.2

It works fine, but with a very unpleasant fact - everyone of the internet can send SIP Register requests to my IP - PBX, so long I have non-stop 'register' attempts from different IP addresses. I tried to do a few firewall access rules, but just RV016 ignores them, when Port-Forwarding rule is applied.

In the previous solution, that I used, it was simple to make a rule to allow an incoming connection on the WAN port only from the IP address of a single source, but, unfortunately, RV016 doesn't have such a feature.

Here, what is the question:

What should I do? I can't leave the situation as it is, but I really want to change the router.

Can someone help me please advice?

Here is an example showing how to add access rules to the top of a port forwarding rule.

When an access rule is set on a port forwarding rule (for example, the SSH service), you want to first add a deny rule to deny all IPs from the side WAN and then add an allow rule to allow a specific IP entering side WAN.

Allow SSH WAN1 [specific IP] [private address]

Deny SSH WAN1 everything [private address]

Tags: Cisco Support

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    * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

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    * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

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    * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

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    * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

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    Can someone help me?

    Thanks in advance!

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    Nadeem

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    To resolve this problem, you will need to add this to jabber - config.Xml.

    true
      msRTCSIP-PrimaryUserAddress
      SIP:

    Kind regards

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  • SIP over VPN and 1.0.2.6 Firmware RV120W

    Updated 1.0.2.6 and all of a sudden devices SIP works via the VPN no longer work. Downgrade from version 1.0.1.3 and they work again. Any ideas? My guess is that some ports are blocked on the VPN in 1.0.2.6

    I thought the whole idea was that fixed bugs rather than introduce firmware ugrades.

    Suggestion for Cisco:-Zip downloads of image of the firmware, or have an upgrade process which includes a CRC check, as it at least the poor punter will have an indication if they have been damaged. I had a subtle memory problem that corrupts certain files. Download of the firmware seems to fill in correctly and you can log on OK but some menu choices resulted in a deadlock with the "Please wait... the page is loading" message. Thorough check of the file sizes revealed that the file I'm downloading in the router is different in size to those on the site, a few hundred bytes must have been corrupted during the download. But the download was normal with no indication of any errors. It's a pretty basic protection measure that should be there as a no-brainer with the router was conducting a CRC check and showing an error if it fails.

    Hello Michael,

    Maybe you have active SIP Application layer gateway. Please try to disable this SIP over VPN works great.

    Firewall--> avancΘs--> remove the checkbox of the SIP ALG.

    Thank you

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  • Problems with SIP scheduling TMS &gt; TP server

    I have a problem of planning where the TMS seems to say my TP server to dial a number to the preconfigured endpoints rooms/external to H323, despite endpoints being configured only with a SIP URI and no ID of H323.

    My external termination points are added as 'rooms' to TMS.  They 'allow reservations' and ' allow incoming SIP URI configured dialing, but all the other slots to be unchecked. "  They have no ID H323, E164, or configured firewall (gatekeeper is set to "off".

    When I have distributed them in a conference, connection settings developed as 'SIP-H323"instead of just"SIP", so he tries H323 numbering first.  It is a problem because many of these external endpoints are CTS-3000 units and if composed as H323, TIP does not work they only connect with a single screen.

    If I manually dial the SIP from the server TP, or if I add an external endpoint to the Conference through TMS and specify SIP, it works very well. It is only a matter because TMS seems to want the MCU try H323 first, despite the configuration 'House' with no option availible H323.

    It just came out recently is because we use TP Server 2.2 in conjunction with the CUBE and manually add endpoints to TP server as "legacy" systems of CTS.  We're heading TP Server 3.0 + VCS - E where TRICK works automatically, but it doesn't seem to work if the call is interoperability of H323, SIP.

    Any ideas how to get MSDS to compose rooms as SIP only (or at least first try SIP)?

    Versions:

    3.0 (2.48) Server TP

    TMS 14.2.2

    VCS 7.2

    Hi Nick,

    I had the same problem with TMS 14.2.2. I have fixed only after you have configured the 'Active SIP server address' field in the configuration page / equipment of the room. You can put any IP address in this field, any.

    It seems that TMS sets TP server to call using SIP only when this field is set to the configuration / the equipment in the room, if I leave this field blank, the result will be the same problem that you are experiencing, a configuration of the connection with H323--> SIP.

    Just to repeat, these are the areas that I have configured in hardware Configuration room to have work with SIP only:

    SIP mode: on

    Active SIP server address: 10.10.10.10

    SIP URI: [email protected] / * /

    Gatekeeper discovery: Off

    Allow the reservation: check

    Allow incoming SIP URI numbering: check

    IP maximum bandwidth: 6000

    You also need to set "Maximum bandwidth IP" field. If this field is '0', you will get an error message "no possible route between participants: TPserver and participating: tests.

    I hope this helps.

    Concerning

    Paulo Souza

    Please note the answers and mark it as "answered" as appropriate.

  • Custom report

    I want to create a customized report detailing Top Conversations for each of my network devices together. However, I want to exclude a certain application.

    Is there a way to filter this particular application? We use an application on our computers that connect to our phones. The SIP application is clogging my report and make the report Top Conversations not as useful as it could be.

    Advice would be greatly appreciated.

    Thank you

    -Evan

    Hey Evan,

    It is impossible to fliter on this request in the report, unfortunately.

    Mario

  • Problem with the commissioning of Web check-in and ASK when swicht to the TMS Provisioning extend mode

    Hi, I need help please, because I have no contract and I cannot open a TAC case.

    I have the following two issues:

    1. when I do the tms extension preparation mode switch as stop working sip calls, I get the following error of internal and internet scenarios for my internal network:

    VCS-e when the call is the Internet to the internal network

    2013-09 - 05T 11: 50:38 - 04:30

    "" "" "" "TVCS: event = 'Search is complete" reason ="authorization not valid - insufficient privilege" Service = "H323" type-aliases-Src ="E164" CBC-alias = '7449"Dst-alias-type ="H323"Dst-alias ="anthony_accardi"call-number ="1a069dfa-1647-11e3-86f9-0010f328943a"Tag ="1a069f44-1647-11e3-b22f-0010f328943a"detail ="found: fake, searchtype:ARQ"Level ="1"elements UTCTime = '2013-09-05 16:20:38, 670"

    VCS - c when the call is internal network to the Internet:

    2013-09 - 05T 11: 53:31 - 04:30

    "" "" "" "TVCS: event = 'Search is complete" reason ="prohibited" Service = "H323" type-aliases-Src ="E164" CBC-alias = '7429"Dst-alias-type ="H323"Dst-alias ="vianyfel_cordaro"call-number ="812a5198-1647-11e3-ba89-0010f325da04"Tag ="812a52e2-1647-11e3-93c9-0010f325da04"detail ="found: fake, searchtype:ARQ"Level ="1"elements UTCTime = '2013-09-05 16:23:31, 687"

    2013-09 - 05T 11: 53:31 - 04:30

    "" "" "TVCS: Event = 'research has attempted" Service ="H323" CBC-alias-type = "E164" CBC-alias ='7429"Dst-alias-type ="H323"Dst-alias ="vianyfel_cordaro"call-number ="812a5198-1647-11e3-ba89-0010f325da04"Tag ='812a52e2-1647-11e3-93c9-0010f325da04" detail = "searchtype:ARQ" Level = "1" elements UTCTime ='2013-09-05 16:23:31, 680"

    2013-09 - 05T 11: 53:23 - 04:30

    "" "" "" "TVCS: event = 'Search is complete" reason ="prohibited" Service = "H323" type-aliases-Src ="E164" CBC-alias = '7429"Dst-alias-type ="H323"Dst-alias ="vianyfel_cordaro"call-number ="7c9181c4-1647-11e3-bda8-0010f325da04"Tag ="7c918304-1647-11e3-865b-0010f325da04"detail ="found: fake, searchtype:ARQ"Level ="1"elements UTCTime = '2013-09-05 16:23:23, 974"

    BUT WHEN THE MODE IS AGENT LEGACY TMS ALL THE CALL WORKS FINE

    2 when I switch I can tms mode of preparation I can do internal network equipment supply but not from the outside and this worries me more is the jabber that being Internet I get the following error:

    013 09 - 05 T 11: 07:42 - 04:30

    "" "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 263"Module ="network.sip"Level = 'INFO': Src - ip ="192.168.0.252"Src-port ="25084"detail = 'receive the Request OPTIONS = method, Request-URI = sip: 192.168.0.250:7001; transport = tls, [email protected] / * /"

    2013-09 - 05T 11: 07:42 - 04:30

    "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level ="DEBUG": Dst - ip ="192.168.0.252"Dst-port ="25084"
    SIPMSG:
    | SIP/2.0 401 Unauthorized
    Via: SIP/2.0/TLS 192.168.0.252:5061; branch = z9hG4bK4de281330ed1277914e57a4bb98ac81416134; received = 192.168.0.252; rport = 25084
    Call ID: [email protected]/ * /.
    CSeq: 38570 OPTIONS
    Starting at: ; tag = 21e96c96b3f9a439
    To: ; tag = ba0e03ca2f6b3957
    Server: TANDBERG/4120 (X7.2.1)
    WWW-Authenticate: Digest realm = "TraversalZone", nonce = "b40cb8278b4a11da992154324161d566d2b57bac3d83c5c518c4528c790d", opaque = "AQAAAN1NC9IHdFS3kNJ3Q6UX2JiBXhut", stale = FALSE, algorithm = MD5, qop = "auth".
    Content-Length: 0

    |

    2013-09 - 05T 11: 07:42 - 04:30

    "" "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level = 'INFO': Dst - ip ="192.168.0.252"Dst-port ="25084"detail ="sending = 401, method = OPTIONS, To = sip response Code: 192.168.0.250:7001, [email protected] / * /"

    2013-09 - 05T 11: 07:42 - 04:30

    "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level ="DEBUG": Src - ip ="192.168.0.252"Src-port ="25084"
    SIPMSG:
    | Sip OPTIONS: 192.168.0.250:7001; transport = tls SIP/2.0
    Via: SIP/2.0/TLS 192.168.0.252:5061; branch = z9hG4bK4de281330ed1277914e57a4bb98ac81416134; received = 192.168.0.252; rport = 25084
    Call ID: [email protected]/ * /.
    CSeq: 38570 OPTIONS
    Starting at: ; tag = 21e96c96b3f9a439
    TO:
    Max-Forwards: 0
    User-Agent: TANDBERG/4120 (X7.2.1)
    Support: com.tandberg.vcs.resourceusage
    Content-Type: text/xml
    Content-Length: 250

    25075024960|

    2013-09 - 05T 11: 07:42 - 04:30

    "" "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level = 'INFO': Src - ip ="192.168.0.252"Src-port ="25084"detail = 'receive the Request OPTIONS = method, Request-URI = sip: 192.168.0.250:7001; transport = tls, [email protected] / * /"

    2013-09 - 05T 11: 07:36 - 04:30

    "" "" "TVCS: elements UTCTime = '2013-09-05 15:37:36, 757" Module ="network.tcp" Level = "DEBUG": Src - ip = "10.10.10.1" Src-port ="10191" Dst - ip = "10.10.10.10" Dst-port ='5060"detail = 'TCP connection is closed"

    2013-09 - 05T 11: 07:36 - 04:30

    "" TVCS: elements UTCTime = '2013-09-05 15:37:36, 641"Module ="network.sip"Level ="DEBUG": Dst - ip ="10.10.10.1"Dst-port ="10191"
    SIPMSG:
    | SIP/2.0 404 not found
    Via: SIP/2.0/TCP 201.210.111.54:2379; branch = z9hG4bK5fc6a3c5021e3557216ef01c2434fb00.1; received = 10.10.10.1; rport = 10191; DefaultZone = ingress-box
    Call ID: [email protected]/ * /.
    CSeq: 301 SUBSCRIBE
    From: <> [email protected] / * />; tag = 2991aa56d191ede3
    To: <> [email protected] / * />; tag = c4114db76ace49d8
    Server: TANDBERG/4120 (X7.2.1)
    WARNING: 200.11.230.253:5060 399 'political response '.
    Content-Length: 0

    |

    2013-09 - 05T 11: 07:36 - 04:30

    "" "" TVCS: elements UTCTime = '2013-09-05 15:37:36, 641"Module ="network.sip"Level = 'INFO': Dst - ip ="10.10.10.1"Dst-port ="10191"detail = 'send = 404, method = SUBSCRIBE, To = sip response Code: [email protected] / * /, [email protected] / * /"

    2013-09 - 05T 11: 07:36 - 04:30

    "" TVCS: elements UTCTime = '2013-09-05 15:37:36, 638"Module ="network.sip"Level ="DEBUG": Src - ip ="10.10.10.1"Src-port ="10191"
    SIPMSG:
    | Sip SUBSCRIBE:[email protected] / * / SIP/2.0
    Via: SIP/2.0/TCP 201.210.111.54:2379; branch = z9hG4bK5fc6a3c5021e3557216ef01c2434fb00.1; received = 10.10.10.1; rport = 10191
    Call ID: [email protected]/ * /.
    CSeq: 301 SUBSCRIBE
    Contact: <> [email protected]/ * /: 2379; transport = tcp >
    From: <> [email protected] / * />; tag = 2991aa56d191ede3
    To: <> [email protected] / * />
    Max-Forwards: 70
    Directions:
    User-Agent: TANDBERG/774 (4.6.3.17194 PCS) - Windows
    Expires: 300
    Event: ua-profile;model=movi;vendor=tandberg.com;profile-type=user;version=4.6.3.17194;clientid="S-1-5-21-1078081533-484061587-725345543";connectivity=1
    Accept: application/pidf + xml
    Content-Length: 0

    The setup I have is:

    Configuration on VCS Expressway:

    TMS Agent Legacy mode

    Search rule:

    local area-no domain

    Any

    Any

    NO.

    Alias matching

    Regex

    (. +) @domain.com. *.

    Replace

    Continue

    LocalZone.GetDaylightChanges

    local area full URL

    Any

    Any

    NO.

    Alias matching

    Regex

    (. +) @domain.com. *.

    Leave

    Continue

    LocalZone.GetDaylightChanges

    Search of covered area rule

    Any

    Any

    NO.

    Any alias

    Continue

    TraversalZone

    Search for DNS zone rule

    Any

    AllZones

    NO.

    Alias matching

    Regex

    (?. *@%localdomains%.*$).*)

    Leave

    Continue

    DNSZone

    Transform

    Transform the alis destinations to URL

    ([^@]*)

    Regex

    Replace

    ------[email protected] / * /

    Presence PUA - on

    Presence server - off

    CONTROL VCS:

    TMS Extension commissioning of fashion

    Search rule

    local area-no domain

    Any

    Any

    NO.

    Alias matching

    Regex

    (. +) @domain.com. *.

    Replace

    Continue

    LocalZone.GetDaylightChanges

    local area full URL

    Any

    Any

    NO.

    Alias matching

    Regex

    (. +) @domain.com. *.

    Leave

    Continue

    LocalZone.GetDaylightChanges

    Search of covered area rule

    Any

    Any

    NO.

    Any alias

    Continue

    TraversalZone

    External IP address search rule

    Any

    Any

    NO.

    Any IP address

    Continue

    TraversalZone

    Transform

    Transform the alis destinations to URL

    ([^@]*)

    Regex

    Replace

    ------[email protected] / * /

    PUA - on

    presence server - on

    I do not have political appeal hace

    Please help me to see what I'm missing or what's wrong?

    Thankss

    Hello

    Ok. Are you saying that VCSe uses the IP address 10.10.10.10 in interface external, right? Of course, what the IP address of 200.x.x.x? It's your VCSe NAT IP address, right? What is this configured in VCSe?

    Well, reaally you have a problem of NAT. look at the SUBSCRIPTION message of jabber to VCSe:

    SIPMSG:

    | Sip SUBSCRIBE:[email protected] / * / SIP/2.0

    Via: SIP/2.0/TCP 201.210.116.201:3612; branch = z9hG4bK138dca6bf6cdd458588900dbaf7b45f4.1; received = 10.10.10.1; rport = 9368

    Call ID: [email protected]/ * /.

    CSeq: 301 SUBSCRIBE

    Contact:

    From: [email protected] / * />; tag = 1e82c817dc3224d5

    In: [email protected] / * />

    Max-Forwards: 70

    Directions:

    User-Agent: TANDBERG/774 (4.6.3.17194 PCS) - Windows

    Expires: 300

    Event: ua-profile;model=movi;vendor=tandberg.com;profile-type=user;version=4.6.3.17194;clientid="S-1-5-21-1078081533-484061587-725345543";connectivity=1

    Accept: application/pidf + xml

    Content-Length: 0

    Do you see? If the Red 192.168.41.205 IP address is the IP address of your router/nat, then you can come to the conclusion that your router is inspection/ALG, it puts its own IP address in the SIP headers. Your router/firewall device should not use any function ALG/inspection, otherwise you will have problems.

    I can say with great confidence, VCSe rejects the message SUBSCRIBE "404 not found" response because VCSE does not recognize this IP address in the field 'road', 192.168.41.205.

    In addition, the configuration of your NAT is not recommended. First, you use the port-based NAT (PAT), in fact, you must use a NAT. Second, when your NAT firewall allows VCSe, the source address is 10.10.10.1, which means that your firewall is NATing the source address and destination address not only. This type of NAT, it is not recommended for h.323/SIP applications.

    Well, don't be angry with me, I try to help, but I need to say, your deployment VCSe is almost completely false, there are a lot of blind spots.

    I suggest reviewing and reconfigure your deployment following this guide:

    http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/VCs/config_guide/Cisco_VCS_Basic_Configuration_Control_with_Expressway_Deployment_Guide_X7-2.PDF

    I hope this helps.

    Concerning

    Paulo Souza

    My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

  • VCSC - questions VCSe

    Hello.

    I have install VCS and VCS Epressway, TMS also control.

    Everything works very well.

    But found that someone could just register on VCSe, he will ask no password and everything.

    If I put DafaultZone or DefaultSubZone to check the credentials, and then customers Movi stops regisering on VCSe.

    In addition, although Movi registers with VCSe, using the windows credentials (when all DefaultZone or DefaultSubZone do not check not the credentials), it is listed as non-authenticated records to the list.

    So, how I don't deny for the unknown customer registration and recording of legmate?

    Hello

    REGISTER request shouldn't be on the format of the URI, but your seem to be:

    SIPMSG:

    | REGISTER sip:[email protected] / * / SIP/2.0

    This means that on one of your RESUME you have a transformation that allows add [email protected]/ * /' incoming SIP requests, and this transformation is the breaking of the REGISTRY.

    You get to have a transformation on your highway that corresponds to ' ([^ @] *)' and this turns------[email protected] / * /? If so, this transformation does not combine with the records transmitted by proxy, and I would recommend you disable this transformation (and consider the consequences in this way).

    This transformation will fundamentally change "REGISTER sip:domain.root" to ' REGISTER sip:[email protected] / * /' which is an illegal syntax for a REGISTER request.

    Hope this helps,

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  • Witz CPL Script problem

    Hello world

    I wrote a script to allow some special calls and reject all other calls.

    Looks like,

    "http://www.Tandberg.net/Cpl-extensions" xmlns: xsi = "http://www.w3.org/2001/XMLSchema-instance" xsi: schemaLocation = "urn: ietf:params:xml:ns:cpl cpl.xsd" > "

    ....................

    My problem is, I need these two rules to allow (matches originally auth.) of SIP and H323 (matches Informati. - origin).

    Why is this? Or I have a problem of anywhere?

    Greetings

    Jens

    Hi Jens,

    My guess would be that the SIP call contains a stated P - a header - them he hits the VCS who this CPL is ongoing, and that authentication to the zone settings, the call is received on is configured as "do not check the credentials. This would lead to an incoming SIP being classed as authenticated, while incoming H323 call would not.

  • TMS - update meetings

    We recently made a change to all of our video systems managed by TMS, allowing to allowing incoming SIP URI composition. I want to update all meetings on hold so that the direction of call setup becomes a reflection of the change. Is this possible?

    Thank you

    RF

    It is not possible to mass update the conference etc. settings in MSD.  However, it may be possible to disable the option of outbound dialing on endpoints and then run "Conference Diagnostics' located under administrative tools > Diagnostics for then automatically fix the mistake of routing created by clearing this option.  If this does not work, you need update entering each conference by one to invoke the call.

  • Purpose of AP_INTERFACE_CONTROLS in the import program open accounts payable.

    Hello
    Can someone explain the purpose of the AP_INTERFACE_CONTROLS table in the open payable import program.

    Concerning
    Deepak

    Hello Deepak.

    AP_INTERFACE_CONTROLS is a temporary table that control separate information in the AP_INVOICES_INTERFACE table when you open Interface import Payables. The table ensures that each import must be unique with respect to the combination of SOURCE and GROUP_ID. This allows multiple imports at the same time.
    Your income Oracle application deletes the information in this table when you perform an import.

    Octavio

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