Re: [OpenSIPS-Users] Which way now?

Russ East Mon, 16 Jan 2012 11:26:26 -0800

Laszlo,

Thank you!  That was it.  Simply added


    $rd = $(du{uri.host});

after calling load_balance() and before t_relay()
in both route and failure_route has fixed the problem.
Wow! A whole lot simpler than what I thought I needed to try next!


Thank you,
Russ





On 1/13/2012 3:33 PM, Laszlo wrote:

U 2012/01/11 21:51:05.620052 xx.xx.xx.23:5060 ->  dd.dd.dd.250:5060
INVITE sip:[email protected]:5060 SIP/2.0.
Record-Route:<sip:xx.xx.xx.23;lr;did=208.986e5031>.
Via: SIP/2.0/UDP xx.xx.xx.23;branch=z9hG4bK5d6.445830c1.0.
Via: SIP/2.0/UDP12.34.56.78:5060  <http://12.34.56.78:5060>.
From:<sip:[email protected]  
<mailto:sip%[email protected]>>;tag=F08DF710-E74.
To:<sip:[email protected]>.


maybe the cisco expects the $rd to be dd.dd.dd.250.

"Beginning with Cisco IOS Software Release 12.4(9)T, administratorscan validate hostnames of incoming initial INVITE messages. When thegateway processes an initial INVITE, a determination is made whetheror not the host portion is in IPv4 format or a domain name.If the host portion is an IP address, its IP address is compared withthe interfaces on the gateway. If a match is found, the INVITE isprocessed as normal. If there is not a match, the gateway sends a 400Bad Request -- `Invalid IP Address' message. "



-Laszlo

2012. január 13. 21:33 Russ East írta, <[email protected]<mailto:[email protected]>>:


    Hi Max,

    Calls are currently coming from 12.34.56.78 and coming into the
    Cisco GW without a proxy.  Why would it not be able to route to
    when a proxy is put in the middle?

    Thanks,
    Russ




    On 1/13/2012 2:21 PM, Max Mühlbronner wrote:

        Hi,

        the response from the Cisco suggests it has a problem with the
        traffic:
        SIP/2.0 400 Bad Request - 'Invalid IP Address'.

        You could try rtpproxy/mediaproxy, it seems like the Cisco GW
        is not able to
        route to the network of the SDP IP (c=IN IP4 12.34.56.78.)?



        BR


        Max M.
        -----Ursprüngliche Nachricht-----
        Von: [email protected]
        <mailto:[email protected]>
        [mailto:[email protected]
        <mailto:[email protected]>] Im Auftrag von Russ
        East
        Gesendet: Freitag, 13. Januar 2012 20:25
        An: [email protected] <mailto:[email protected]>
        Betreff: [OpenSIPS-Users] Which way now?

        Hello guys.

        I've been trying to avoid asking for some advice, but three
        weeks into
        studying, watching the webinars, reading the forums,
        mail-lists, and testing
        I've reached a point of "Now What!". I can't find a clear path
        in the
        documentation for what I am trying to do, even though the task
        seems very
        simple!

        I put in a second Cisco 5400 for our local DID numbers (Direct
        Inward Dial -
        not dialog ID) from the PSTN over VoIP.  The PSTN will only
        send to one IP
        and will not accept redirects, so I though I just needed a
        "simple" proxy
        and load balance to the ciscos.  Seven or eight years ago I
        helped a friend
        setup openSER serving two asterisks - no problem.  But I am
        having problems
        here going to the ciscos and am trying to find which
        combination of modules
        I need to use...

        I believe I need to go to topology hiding and a media proxy.
         Do I use
        rtp_proxy or something else like MediaProxy?

        Here's what I want to accomplish... I need to successfully
        send the calls to
        the ciscos, balanced.  All are public IP addresses.

        Attached is a sample of my most recent failed attempt with
        call scenario
        data, mysql data, route script and sip trace.

        Any advice or examples would be appreciated!


        Thank you,
        Russ



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