Disable SIP ALG in RV042 router?

Using these routers for VoIP phones, the manufacturer says that SIP ALG must be disabled. I cannot find a setting for this.

RV042 does not suport SIP ALG.

Similar Messages

  • DDR2200 Disable SIP ALG

    Hello;
    I am attempting to assist a friend who was given a DDR2200 by their ISP, Centurytel.
    We are trying to disable the SIP ALG in the gateway but the link is not present when we log into the gateways GUI.
    We've tried to get Centurytel to assist us but all they have done is try to update the firmware.
    software version v00.00.03.45.4e
    hardware version v06
    When we show them where the SIP ALG should be located in the GUI when referencing the DDR2200 user manual (pg 63) but they are not able to explain why we cannot see the link and are unable to assist us further.
    I need to disable the SIP ALG as it's interfering with a SIP based Audio/Video Chat application my friend uses.
    Any assistance would be greatly appreciated?
    Thanks in advance.
    Posted by WebUser Matthew Wichlan

    Hello.
    The standard port for SIP is port 5060, so Cisco runs an ALG on this port.
    You've switched it off, so there os NO ALG running for UDP at least.
    Non standard ports won't be running the SIP ALG by default, although you can switch one on using the command you found.
    Adam

  • RV016 SIP ALG

    Hi 
    Since 2 week I have some problem with my voip provider .
    In my pabx software (3cx) my voip provider is correctly register but sometimes i dont know why I can't received external to internal call. 
    When i call my voip provider he tell me to disable sip alg of my router but in my RV016 i don't find any option sip alg.
    How must i do?
    My voip provider tell me to do the command line 
    no ip nat service sip udp port 5060
    in telnet but i search and i can't acces telnet because i don't know login/password 
    Thanks for your help
    Best Regard 
    Loic

    Hi Loic,
    Can you try please from the browser :  https://IPaddress_of_rv016/f_general_hidden.htm
    You will see the SIP ALG option
    Please rate the post or mark as answered to help other Cisco Customers
    thanks
    Mehdi

  • WRV210 SIP ALG disable?

    Can i disable the wrv210 sip alg ? it messes with my voip provider and must be disabled...i use a pap2T adapter and it works fine for outgoing calls...but after a while incoming calls half ring and i cant answer them.

    please look at this link.....it doesn't state the info on the SIP ALG everyone you look at info on the WRV210...but this clearly states it has it built in to support voip to help traverse the firewall.....but it does it behind the scenes and cannot be turned off
    http://www.cisco.com/en/US/prod/collateral/routers/ps9923/ps9929/data_sheet_c78-502735.html

  • Sip phone behind RV042

    Hi All
    We have recently installed an RV042 in place of a dead Zyxel router. The user has 1 Gigaset sip phone on site which registers. The issue we seem to have is after a who the phone will then show as unregistered and if we reboot the router all is fine again for a while.
    I have made sure the firmware is up to date and there is no sip alg option to disable. It's driving me around the bend trying to reolve it. If I can't get this fixed it blows out using this as a replacement for other customers
    Any advise here would be great
    Thanks
    Glenn

    Hi,
    In RV042 SIP ALG option is available in hidden page.Use the below link to access the hidden page of RV042 to disable the SIP ALG option.
    https://192.168.1.1/f_general_hidden.htm
    regards
    Moorthy

  • Enabling the sip/alg

    Hi 
    I am looking to use a softphone from a third party software and for that  I have just purchased a new BT Home Hub 3 in the hope that it would work (I had BT Home Hub 2 and was told to buy the Home Hub 3) but unfortunately it doesn't.
    I was told that ONLY BT can disable my sip/alg and for that I phone the Technical support about 6 times and nobody seems to know what I am talking about. Now, I am with a second router and I cannot get what I am looking for, I am so frustrated.
    Does anyone know how I could get the sip/alg disabled?
    Many thanks,
    Karine 

    You should be able to run it on either. If you have a choice, it would probably be better to run it on the upstream router. There are some things with SIP that aren't supported with the NAT ALG when the SIP is originated and fixed on the same router. One of these things I can name off hand is the 'sip bind' command which isn't supported on the same box.
    It's still very possible to run it on the same box, however. But the SIP ALG doesn't require any information from CME to work properly.

  • SIP ALG feature queries

    I have few queries related to SIP ALG feature.
    - Is SIP ALG(NAT) and SIP inspection(inspect sip) dependent on each other? Do we need to enable both features on a router? Or are they independent?
    My understanding is both are separate features and SIP ALG make sure it open up doors on NAT for any incoming connections from outside using the ports negotiated. And SIP inspection(the 'inspect sip' command) can be added for security purposes to make sure the router when acting as a firewall dynamically allows ports for traversal.
    Please correct if my understanding is correct.
    - Why do docs/discussions mention SIP ALG feature (enabled by default) breaks communication(one way/disconnects)? Can someone illustrate why would this feature result in a call breaking?
    - Is there a difference in these two features in the way it is implemented on an ASA vs routers?
    Would be grateful if someone could help on my queries. Tried searching in internet but didn't find clear info.
    Thanks,
    Karthic

    Hello, 
    It has been always supported on the ASA platform. You just need to turn on the SIP inspection.
    Mike 

  • CUBE and NAT without SIP ALG

    I was wondering if anyone had a CUBE SIP Profile example for rewriting SDP to fix private-to-public IP address in the SDP so that CUBE can be used behind a static NAT without SIP ALG.

    Im trying to connect a SIP softphone (on the outside) to a IP PBX on the inside.  I am seeing postings that say that "ip nat service sip" is the command that enables that feature, and others say that it breaks it.  So far my testings shows that it does break it.  Ultimately I want my outside softphone to register to the Phone system as an external IP address.  It seems like SIP normally relays the internal IP address and the ALG router will make the translation on outbound and send it to the right source. 

  • SIP ALG / SIP NAT Traversal

    I have a 2900 series router running IOS version 15.1(4).  I am trying to connect 3rd party sip softphones to a 3rd party SIP Call controller on the inside.  With low-cost firewall/gateways, I normally enable the SIP ALG feature and it will dynamically open the UDP ports for a SIP conversation for the duration of it and then close them.  Does cisco IOS firewall have a SIP ALG feature and how do I configure it?  Any guidance is much appreciated. 
    Eddie

    Im trying to connect a SIP softphone (on the outside) to a IP PBX on the inside.  I am seeing postings that say that "ip nat service sip" is the command that enables that feature, and others say that it breaks it.  So far my testings shows that it does break it.  Ultimately I want my outside softphone to register to the Phone system as an external IP address.  It seems like SIP normally relays the internal IP address and the ALG router will make the translation on outbound and send it to the right source. 

  • Does Airport Time Capsule have SIP ALG

    I have an Airport Time Capsule and am trying to use my Bria 3 App for my company VoIP service, My IT guys have asked if the Airport Time Capsule has SIP ALG turned on.  I have'nt a clue where to look, any ideas?

    SIP and ALG are features that would be found on commercial grade routers.
    The Time Capsule has no settings for VoIP. It is a very simple home router with a built in hard drive.

  • SIP ALG for NAT Cisco 3845

    Does the SIP ALG for NAT have to be on the router where I am planning on running Call Manager Express for that to work properly, or can I have it running on an upstream router where I am currently preforming NAT functions?

    You should be able to run it on either. If you have a choice, it would probably be better to run it on the upstream router. There are some things with SIP that aren't supported with the NAT ALG when the SIP is originated and fixed on the same router. One of these things I can name off hand is the 'sip bind' command which isn't supported on the same box.
    It's still very possible to run it on the same box, however. But the SIP ALG doesn't require any information from CME to work properly.

  • Please help with SIP configuration on 2801 router

    Hi All.
    Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.
    The information from our SIP provider:
    We have issued the following DDI range: 018877000 – 99
    There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.
    Configuration details are as follows:
    Our Primary Proxy:-        99.234.56.78
    Codec supported:-             G711Alaw, G729 (G711Alaw is the preferred codec)
    Fax Support:-                     T38 and G711Alaw
    DTMF:-                                 RFC2833 and INFO
    CLI Method:-                     Remote-Party-ID
    Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.
    This is a SIP configuration on Cisco2801 router (I used outgoing calls only):
    translation-rule 10
    Rule 0 ^90 0
    Rule 1 ^91 1
    Rule 2 ^92 2
    Rule 3 ^93 3
    Rule 4 ^94 4
    Rule 5 ^95 5
    Rule 6 ^96 6
    Rule 7 ^97 7
    Rule 8 ^98 8
    Rule 9 ^99 9
    interface FastEthernet0/0.1
    description ***DATA VLAN***
    encapsulation dot1Q 1 native
    ip address 10.1.1.101 255.255.255.0
    interface FastEthernet0/0.2
    description ***VOICE VLAN***
    encapsulation dot1Q 2
    ip address 192.168.22.1 255.255.255.0
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    h323
      call start slow
    sip
      bind control source-interface FastEthernet0/0.2
      bind media source-interface FastEthernet0/0.2
      registrar server expires max 36000 min 600
    voice class codec 1
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    dial-peer voice 1 pots
    description ### External Dialling via BRI ###
    preference 7
    destination-pattern 9T
    translate-outgoing called 10
    direct-inward-dial
    port 0/0/0
    forward-digits all
    dial-peer voice 2 pots
    description ### External Dialling via BRI ###
    preference 2
    destination-pattern 9T
    translate-outgoing called 10
    direct-inward-dial
    port 0/0/1
    forward-digits all
    dial-peer voice 9000 voip
    description ** Outgoing calls to SIP **
    preference 1
    destination-pattern 9T
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target ipv4:99.234.56.78:5060
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    sip-ua
    timers connect 100
    sip-server ipv4:99.234.56.78
    I used debugging commands to troubleshoot the calls.
    2801(config-dial-peer)#
    094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018
    094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
    094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9
    094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH
    094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90
    094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH
    094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=908
    094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH
    094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9086
    094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH
    094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862
    094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH
    094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=908621
    094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH
    094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9086215
    094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH
    094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157
    094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH
    094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=908621577
    094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
    094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=9086215777
    094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Partial Matches(1) after DP_MATCH_DEST
    094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=MORE_DIGITS_NEEDED(1)
    094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
    094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774T
    094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
    094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL
    094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:10 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397230
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:11 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397231
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:12 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397232
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:14 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397234
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    I made some changes in the router configuration.
    I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
    The debugging is changed now. I can send and receive a respond from SIP server. But  It shows an error: SIP/2.0 404 Not Found
    Then it moves to ISDN line, and use this line to make a call.
    102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
    102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774T
    102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
    102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
    103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
    Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 14:45:47 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327416347
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 19412 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 404 Not Found
    From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
    To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
    Content-Length: 0
    103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
    From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
    To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
    Date: Tue, 24 Jan 2012 14:45:47 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
    103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
    103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=211
    103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=20018
    103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=0862157774
    103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
    103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
       Result=NO_MATCH(-1)
    103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
    2801(config-dial-peer)#
    Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
    But it didn’t affect anything.
    Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
    Really stuck and don't know where to look at.
    Any help will be highly appreciated.
    Thanks.

    Hi Dan.
    Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
    I use Cisco ASDM for ASA to make changes.
    There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44)  for a few ports.
    Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
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    116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
    116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
    116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:25 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505305
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:26 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505306
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:27 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505307
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:57 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
    User-Agent: Cisco-SIPGateway/IOS-12.x
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    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505337
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    I'll add Incoming dial-peer now.
    Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
    Appretiate your help.
    Thanks a mill.

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