RV220W sip problem

I have installed 2 RV220w routers and 5 RV120w routers on a 7 site vpn ( all with the latest 10.5.8 furnware)
I am running an Elastix 2.4 asterisk based pbx
The issue I am having is this:
With the sip alg disabled and the port forward set for 5060 udp and 10000-20000 udp I cannont recieve sip calls
With the sip alg enabled and the port forward set for 5060 udd and 10000-20000 upd I can reiceive calls BUT
the rtp packets are coming at any random port 9000 to 50000
Aparently the sip alg is rewriting the sip header causing this
Normally I would run this with the sip alg turned off but in that mode I cannot get the sip port to pass calls at all
I am thinking of downgrading the firmware to the 10.4.17
Does anyone have experience with this
Thanks

Please see the following thread:
https://supportforums.cisco.com/thread/2269123
- Marty

Similar Messages

  • E61: sip problem with Gizmo, cannot make call

    I register for Gizmo few days ago. I cannot make any call out (to 411, 1-747xxx numbers, any landline, any mobile )
    I cannot hear the other side, but the other side hears my voice
    what's wrong?
    I'm using the latest v3 firmware

    I have some problems with Gizmo as well. Not exactly the same problem as yours. In my case, initiating a call takes too long (~1 minute). I would recommend trying Truphone. It is another SIP-based app; and in my opinion, much better than Gizmo.
    You can also check my blog posting here, http://www.s60tips.com/2007/06/28/which-voip-applications-to-use-part-v/Message Edited by antonypranata on 05-Sep-200709:22 AM
    Antony Pranata
    Visit S60Tips.com for tips, tricks and tutorials of using S60 phones

  • RV220W PPPoE problems

    Hi I am just trying to setup an RV220w on a FTTC BT Openreach modem and am finding that the router will not establish a connection.
    In the logs I have seen the following:
    Sat Jan  1 00:06:21 2000(GMT) [rv220w][System][PLATFORM] pppoeMgmtTblHandler: pppoe enable failed
    Sat Jan  1 00:06:21 2000(GMT) [rv220w][System][PLATFORM] pppoeMgmtDBUpdateHandler: error in executing database update handler
    I have seen some issues that people are having with the 1.0.1.0 firmware, in particular PPPoE issues. I note the router came preinstalled with 1.0.1.0 so have downgraded to 1.0.0.26.
    I have removed the settings and tried again, but receive the same error message above.
    The PPPoE settings are as follows:
    username:               [******]@[**********].[***]
    password:               [******************]
    Authentication Type:    Auto-Negotiate
    At a bit of a loss what is going on ? I have tried resetting it to factory defaults on firmware 1.0.0.26 and entered the details again, but have the same error.
    Has anyone else had success on this, I have other PPPoE devices that work absolutely no problems?

    Have you tried installing the Visual C++ Redistributable 2005?  On 64-bit machines install both the 32-bit and 64-bit versions.

  • UCCE 9 Outbound SIP problem

    Dear All,
    I'm implementing the SIP Outbound Dialer, the architecture is below
    1. ROGGER + Campaign Manager : (version 9.0)
    2. CMPG + CTI Server + CTIOS + MRPG + SIP Dialer : (version 9.0)
    3. CUCM ver 8.6
    4. Gateway + E1 trunk + IOS 15.1(3)T
    The problem is SIP Dialer doesn't call out after successfully loaded calling list into the CampaignManager (checked the log in baImport). I checked the status of SIP Dialer that it Active all (please, see the attach file). But when I check the log on MRPG it said that "Failed an attempt to ACTIVATE the Peripheral's Routing Client ". I don't sure that this is the cause of the problem or not, I've re-checked configuration on MRPG manay times but nothing strange. Anyone found this problem before please, suggest.
    BR.
    Winai K.

    Hi,
    Except for 15.1(x)T train other IOS versions do not have the capability of doing the CPA analysis (determining voice \answering machine \ fax etc) effectively.
    As a result it is recomended to use above code.
    Thank you
    Anuj

  • RV220W Website Problems

    I am new to networking so I will do my best to express the problem I am having:
    We replaced our router last weekend with the RV220w and ever since we did that we have problems getting to external websites.  The most ironic one would be cisco.com, we can't get to it.  We get a Server Not Found on our web browser(s).  I've tried Firefox 7 and 8, IE 8 and 9 and Safari.
    Other issue we are having is that sometimes when we do get to a site the page is loaded wrong, not sure how to explain this one.  It is like it displays all the text but doesn't display the structure of the site (no color, no background image, no tables, etc, etc)
    I've tried firmare 1.0.2.4 and 1.0.3.5 and we have the same result.  I've disabled PPTP server which I have read in many forums causing other problems.
    Any ideas?
    Michael

    Hi Michael,
    This sounds like an MTU issue. (Packet fragmentation) If you are using DSL, go to Networking-> WAN-> IPv4 WAN Configuration and at the bottom of the page change the MTU to custom. Lower the MTU in increments of 10, starting at 1492. Usually you will find that the pages start loading properly after you lower the Maximum Transmission Unit. To find the correct MTU to use without guessing you can do an MTU test.
    http://help.expedient.com/broadband/mtu_ping_test.shtml
    The value that you get after adding 28 is what you input in the RV220W.
    Please keep us updated.

  • SIP problem port

    Hi,
    I'm trying to send a REGISTER Sip Message to Asterisk with credentials (following the instructions from Javadoc of JSR180). The code is the follow:
    try {
    // open listener in application specific port 5080
    sipNotifier = (SipConnectionNotifier)Connector.open("sip:5080");
    // build the contact URI
    contact = new String("sip:emanuele@"+sipNotifier.getLocalAddress()+":"+sipNotifier.getLocalPort()+";transport=UDP");
    // open client connection to the SIP registrar in this case "host.com"
    sipConnection = (SipClientConnection) Connector.open("sip:89.89.89.89");
    sipConnection.setListener(this);
    // initialize REGISTER with appropriate headers
    sipConnection.initRequest("REGISTER", null);
    sipConnection.setCredentials("ciccio", "ciccio", "mydomain.com");
    sipConnection.setHeader("From", "sip:[email protected]");
    sipConnection.setHeader("To", "sip:[email protected]");
    sipConnection.setHeader("Expires", "600");
    sipConnection.setHeader("Contact", "<"+contact+">");
    sipConnection.send();
    boolean handled = false;
    int scode = 0;
    while(!handled) {
    // wait max 30 secs for response
    sipConnection.receive(30000);
    scode = sipConnection.getStatusCode();
    switch(scode)
    case 200:
    // handle OK response
    handled = true;
    break;
    default:
    // handle other responses
    handled = true;
    // wait maximum 15 seconds for response
    } catch ( IOException e ) {
    e.printStackTrace( );
    voipmidlet.addMessage( "Error: " + e.getMessage( ) );
    finally
    if (sipConnection != null)
    try {
    sipConnection.close();
    catch ( IOException e ) {
    e.printStackTrace();
    Analyzing the traffic I can see the server receives the first register message, answers with unauthorized but my j2me app doesn't catch this response.
    The server respond on the same port that is the source port the client send message from and that port is chosen randomly at the moment of the send and it's different from the port my listener is where.
    Does anyone know a solution?
    Thanks
    Emanuele

    Sorry I have the lines:
    sipConnection.initRequest("REGISTER", sipNotifier);
    and not
    sipConnection.initRequest("REGISTER", null);
    in my code....the code in the previous post has that error but also with sipNotifier it doesn't work due the same problem.

  • RV220W connection Problem

    Hello,
    i have a problem with my  rv220w router.
    - DHCP request are not always answered
    - ping's in a network take part between 200-500 ms
    - Is after the router a switch, devices are behind the switch do not work sometimes. No DHCP is working.
    On the router is the actual firmware.
    have you any idea?
    ps: Please excuse my english.

    Hi Martin, thank you for using our forum, my name is Johnnatan I am part of the Small business Support community. I apologize for your issue you are having, I was wondering about your firmware version, just ensure you are running the current version (1.0.4.17)if not  you can download it here
    If you already are running the correct firmware you could check the MTU size, in this case I advise you to decrease or increase in order to find the correct size for your device.
    If the issue continues you can perform a factory reset and configure the device manually just the basic configuration, because it looks like a low performance issue. I hope you find this answer useful
    “Please rate useful posts so other users can benefit from it”
    Greetings, 
    Johnnatan Rodriguez Miranda.
    Cisco Network Support Engineer.

  • SIP problem with Huawei b593....router or EE4G PAYG?

    I'm struggling to get any SIP service to work on my Huawei b593s-22 / EE4G Data PAYG sim.  The SIP/VOIP service hangs on 'registering' and I can't make outgoing calls.  If I put in my old Three data sim, the SIP/VOIP go to UP status immediately, so it must be EE - right?  A quick search tells me VOIP isn't blocked on EE PAYG?? Freespeech support told me "This is the source of your problem, the IP address 100.108.207.89 is not a public IP. This means your provider EE 4G is placing a NAT (to translate the private IP 100.108.207.89 to a shared public one) between you and the public internet. This is OK for web surfing but will not work for VoIP. You need to have a public IP address assigned. Talk to EE 4G and see if they have a solution for you." Any suggestions?

    Hi,
    I'm also having difficulties. I have enabled the UPnp, but the Back to My Mac details box is still stating that on my Huawei LTE CPE B593, NAT-PMP is turned off.
    Any ideas?
    Thanks
    John

  • N95 V20.0.15 - SIP Problem

    Sip registration fais after upgrade to v20.0.15. Worke fine with previous version v12.
    Anyone has similar problem?
    Is their any solution or we have to wait for the new version

    Registration works fine for me, BUT if I receive a SIP call from a client with enabled video the phone hangs completely (I have to remove battery, no workaround).
    This didn't happen with v12.
    This is very annoying, because since I have no control over the settings of the softphones used by the people that call me, having my phone registered to Internet Telephony service offers a great risk of leaving me aout of service.
    Is there any solution to this?
    Thanks
    Jorge

  • Error -8, SIP problems...

    Hello everyone!
    I am having some problems trying to video chat with iChat.
    Me and the other end are using iChat AV 3.1.8 (v448) on OS X Tiger 10.4.9.
    We both can connect to the Apple test bot (AIM Users appleu3test01, appleu3test02, appleu3test03) but when we try to connect to each other we get the -8 error, where sometimes the SIP (Seesion Initiation Protocol) comes.
    My question is: Is there any solution for this error?
    Thanks anyway,
    Andre Taniguti

    Hi
    First have you both set the Quicktime streaming setting, goto sys prefs/quicktime/streaming/streaming speed, set to 1.5mbps(dont use automatic)
    In ichats prefs click on video and change bandwidth limit to NONE.
    Restart iChat.
    Tony

  • ZBFW and SIP problems

    Hi GUYS,
    Please help me..
    I have experiencing problems with SIP phones behind firewall running on CIsco 887 VA-M.
    I got these messages :
      5 02:43:37.439: %AIC-4-SIP_PROTOCOL_VIOLATION: SIP protocol violation (Mandatory header field missing) -  dropping udp session 192.168.33.120:5061 203.111.37.20:5060 on zone-pair in-out-zone class cmap-in-out-base
    Jul  5 02:43:40.035: %AIC-4-SIP_PROTOCOL_VIOLATION: SIP protocol violation (Mandatory header field missing) -  dropping udp session 192.168.33.117:5060 203.111.37.20:5060 on zone-pair in-out-zone class cmap-in-out-base
    I have downgraded software to 151-4.M6 and greated the policy to skip those checkings but no any improvements
    My config is
    boot-start-marker
    boot system flash:c880data-universalk9-mz.151-4.M6.bin
    boot-end-marker
    no aaa new-model
    memory-size iomem 10
    crypto pki token default removal timeout 0
    ip source-route
    ip dhcp excluded-address 192.168.33.1 192.168.33.99
    ip dhcp excluded-address 192.168.33.150 192.168.33.254
    ip dhcp pool 1
    network 192.168.33.0 255.255.255.0
    default-router 192.168.33.1
    dns-server 8.8.8.8
    ip dhcp pool `
    ip cef
    ip domain name ues
    ip name-server 8.8.8.8
    no ipv6 cef
    license udi pid CISCO887VA-M-K9 sn FGL171725DT
    controller VDSL 0
    class-map type inspect match-all cmap-manage
    match access-group 23
    class-map type inspect match-any cmap-in-out-ALL_allowed
    match access-group 150
    class-map type inspect match-any cmap-in-out-base
    match protocol https
    match protocol http
    match protocol dns
    match protocol ftp
    match protocol pop3
    match protocol citrix
    match protocol citriximaclient
    match protocol icmp
    match protocol smtp
    match protocol pptp
    match protocol gopher
    match protocol sip
    match protocol h323
    match protocol sip-tls
    policy-map type inspect allow_all
    class type inspect cmap-in-out-ALL_allowed
      pass
    class class-default
      drop
    policy-map type inspect pmap-out-in-manage
    class type inspect cmap-manage
      pass
    class class-default
      drop
    policy-map type inspect pmap-in-out
    class type inspect cmap-in-out-base
      inspect
    class type inspect cmap-in-out-ALL_allowed
      pass
    class class-default
      drop
    zone security in
    zone security out
    zone-pair security in-out-zone source in destination out
    service-policy type inspect pmap-in-out
    zone-pair security out-self-zone source out destination self
    service-policy type inspect pmap-out-in-manage
    zone-pair security out-in-zone source out destination in
    service-policy type inspect allow_all
    interface Ethernet0
    no ip address
    shutdown
    no fair-queue
    interface ATM0
    no ip address
    no ip route-cache
    load-interval 30
    no atm ilmi-keepalive
    pvc 8/35
      encapsulation aal5mux ppp dialer
      dialer pool-member 1
    interface FastEthernet0
    switchport access vlan 100
    no ip address
    interface FastEthernet1
    switchport access vlan 100
    no ip address
    interface FastEthernet2
    switchport access vlan 100
    no ip address
    interface FastEthernet3
    switchport access vlan 100
    no ip address
    interface Vlan1
    no ip address
    interface Vlan100
    ip address 192.168.33.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly in
    zone-member security in
    interface Dialer0
    ip address negotiated
    no ip redirects
    no ip unreachables
    no ip proxy-arp
    ip mtu 1492
    ip flow ingress
    ip nat outside
    ip virtual-reassembly in
    zone-member security out
    encapsulation ppp
    ip tcp adjust-mss 1350
    dialer pool 1
    ppp authentication chap pap callin
    ppp chap hostname
    ppp chap password 0 673569
    ppp pap sent-username
    no cdp enable
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip nat inside source list FOR_NAT interface Dialer0 overload
    ip route 0.0.0.0 0.0.0.0 Dialer0
    ip access-list extended FOR_NAT
    permit ip 192.168.33.0 0.0.0.255 any
    ip access-list extended KILL-TFTP
    deny   udp any eq tftp any
    permit ip any any
    access-list 150 permit ip any any
    access-list 150 remark TEMP
    line con 0
    no modem enable
    line aux 0
    line vty 0 4
    login local
    transport input ssh
    end
    Thanks a lot!

    Try to do disable inspection of protocol-violation for sip, using this config:
    class-map type inspect sip SIP_VIOLATION_CLASS
    match protocol-violation
    policy-map type inspect sip SIP_VIOLATION_POLICY
    class type inspect sip SIP_VIOLATION_CLASS
    allow
    policy-map type inspect pmap-in-out
    class type inspect cmap-in-out-base
      inspect
      service-policy sip SIP_VIOLATION_POLICY

  • SIP problem to place calls

    Hello,
    We are a SIP provider in France. More and more of our customers are using the WIFI/SIP features of Nokia mobile phones. They can register without problem, as well as they can get SIP calls on their mobile.
    Yet, many of them have problems to place calls. As far as we can see in the SIP traces, it looks like the N95 answers by a CANCEL to a 180 Ringing message.
    We have done some tests with a customer with the following configuration :
    Nokia N95 8GB
    V 20.0.16 28-02-08 RM-320
    Is this a known problem ? Would it be possible to get in touch with some developers of the SIP stack to trace this problem ?
    Thanks and regards,
    Guillaume
    Solved!
    Go to Solution.

    I wouldn't be so sure of that. I have an N95-1 registered to my own Asterisk server and I can place calls no problem.
    This said, if you want to get hold of Nokia you've come to the wrong place. This is just a forum for users of Nokia products to share information. You should be able to contact a Nokia customer service representative on 0811.004567 and they should be able to pass the message on.
    Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

  • SIP Problem - Can make, but not receive calls

    I two E-Series phones (E51 and E71). The E71 is configured to connect over my WLAN to my corportate phone system. this works perfectly for making and receiving internal and external calls, however, my E51 can only make calls. both phones have identical settings with regard to my phone system but i wonder if there is a known bug and more importantly a workaround? i have also tried to configure a spare E65 and this presents the same problem as the E51.
    Any ideas? starting to get a headache over this....
    Many thanks
    Ben

    I am sorry I didn't get back to you but I have been away for a while.  I have discovered that when I turn the firewall off on my imac I can receive Facetime calls - turn it back on and I can't.  Seems a little silly that Apple's own firewall blocks Facetime.
    I wonder if anyone else has this problem.  I am a little concerned about having to keep my firewall turned off.
    Thank you very much for your help anyway.

  • SIP problem

    I am angry, i have tryed hard to get my E72 SIP-settnings working steady, but not. It just doesent work, after a while it wont register, and than it lock "SIP setting" so i can not access it, I have to hardware reset and make the installation again (ALL installations, contacts, programs etc) Its very very frustating. After new installation I can create a new SIP and it works shortly again.
    "SIP-settings" is allready in the phone 
    To access "Advanced VoIP"  I downloaded an installed "SIP VoIP 3.x"
    (Look here:  http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...  (The other  programs there is not comatible)
     But as i said earier, it just doesent work. I have read a lot here on this forum, and around internet "Step by step" instructions but just doesent work.
    Sometimes it works a couple of times, but never stable and after a while not at all.
    What to do?? Somone have any ideas? Is there realy somone who have successed with this?  If plese make an step by step instruction! I think I have spend at least 14 hour trying to fix this....:-((

    I had always trouble with nokia SIP and voip settings , and getting it to work.
    Use a sip client application, like Nimbuzz or fring.
    Much easier setup and works straight.
    Haikal

  • Nokia N79 SIP Problem

    I've just purchased my new N79 and configured the SIP setting using  Tools > Settings >Connection > SIP settings.
    everything went smoothly and my sip account is registered successfully. After this i tried to find out the Internet Tel option as nokia E65 has this option but it is not there in N79. I don't get the option intenet call when i select options>call.
    Kindly anyone knows how to do internet call afer SIP settings are successful??
    Regards
    Inam 

    Hi Amit here,
    Go to this website http://www.forum.nokia.com/info/sw.nokia.com/id/b1c361a2-7eb2-4853-8c0c-d2f54e184237/SIP_VoIP_3_1_Se...
    First you have to make your account on nokia forum. 
    Download version 3 software and install on your phone.
    Then go to menu-tools-connectivity-net settings-advance VOIP setting and make a new profile using your SIP profile.
    Then it will go to contact automaticaly or if now going go to cantact and you will find there Activate service.
    After activating this service go to any contact and option-call-internet call
    Thank you

Maybe you are looking for

  • Display Port - 50" Full HD Plasma, what adapter?

    Hey guys I have this 50" TV, and ofc I want to hook it up with the MB, but how to do it? There is two adapters: Mini DisplayPort to DVI Adapter. "....DVI adapter lets you connect an advanced digital monitor, such as the 20- or 23-inch Apple Cinema Di

  • Support for Exchange/MessageMirror in Mail app -

    is there any way to give the native mail application access to a MessageMirror mailbox on an Exchange server?  Based on what I can find, the answer is "no," because the mail application doesn't support delegated mailboxes.  I have two Exchange accoun

  • Missing air play icon

    I have a late 2011 macbook pro.  The icon for air play is showing up in my itunes but not on my main tool bar.

  • Open pdf-file on mac OS 10.4

    Does anybody know how to open a pdf-file on mac OS 10.4 (tiger, jvm 1.5). I already saw related links but couldn't find the clue.

  • Need some suggestions on ABAP webdynpro ..

    Hi I am actually a java guy. But now i m moving to Object Oriented ABAP. I have to work in ABAP Webdynpro. Could anyone please tell me How much time it will take to learn OOAbap ? Could u please suggest me which book is best for learning OOABAP ? Any