[OpenSIPS-Users] Which way now?

[Russ East]

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:100012121112222 at xx.xx.xx.23:5060 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:12125551212 at 12.34.56.78  <mailto:sip%3A12125551212 at 12.34.56.78>>;tag=F08DF710-E74.
> To:<sip:100012121112222 at xx.xx.xx.23>.
>
>
> maybe the cisco expects the $rd to be dd.dd.dd.250.
>
> "Beginning with Cisco IOS Software Release 12.4(9)T, administrators 
> can validate hostnames of incoming initial INVITE messages. When the 
> gateway processes an initial INVITE, a determination is made whether 
> or 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 with 
> the interfaces on the gateway. If a match is found, the INVITE is 
> processed as normal. If there is not a match, the gateway sends a 400 
> Bad Request -- `Invalid IP Address' message. "
>
>
> -Laszlo
>
>
>
> 2012. január 13. 21:33 Russ East írta, <russ at omahapoolplayers.com 
> <mailto:russ at omahapoolplayers.com>>:
>
>     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: users-bounces at lists.opensips.org
>         <mailto:users-bounces at lists.opensips.org>
>         [mailto:users-bounces at lists.opensips.org
>         <mailto:users-bounces at lists.opensips.org>] Im Auftrag von Russ
>         East
>         Gesendet: Freitag, 13. Januar 2012 20:25
>         An: users at lists.opensips.org <mailto:users at lists.opensips.org>
>         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
>
>
>
>         -----
>         eMail ist virenfrei.
>         Von AVG überprüft - www.avg.de <http://www.avg.de>
>         Version: 10.0.1416 / Virendatenbank: 2109/4140 - Ausgabedatum:
>         13.01.2012
>
>
>         _______________________________________________
>         Users mailing list
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20120116/863be2a4/attachment.htm>