<pre>U 2012/01/11 21:51:05.620052 xx.xx.xx.23:5060 -> dd.dd.dd.250:5060
INVITE sip:100012121112222@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/UDP <a href="http://12.34.56.78:5060">12.34.56.78:5060</a>.
From: <<a href="mailto:sip%3A12125551212@12.34.56.78">sip:12125551212@12.34.56.78</a>>;tag=F08DF710-E74.
To: <sip:100012121112222@xx.xx.xx.23>.<br><br><br></pre>maybe the cisco expects the $rd to be dd.dd.dd.250.<br><br>"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. <br>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. "<br><br><br>-Laszlo<br><br><br><br><div class="gmail_quote">2012. január 13. 21:33 Russ East írta, <span dir="ltr"><<a href="mailto:russ@omahapoolplayers.com">russ@omahapoolplayers.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Max,<br>
<br>
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?<br>
<br>
Thanks,<br>
Russ<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
<br>
On 1/13/2012 2:21 PM, Max Mühlbronner wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
<br>
the response from the Cisco suggests it has a problem with the traffic:<br>
SIP/2.0 400 Bad Request - 'Invalid IP Address'.<br>
<br>
You could try rtpproxy/mediaproxy, it seems like the Cisco GW is not able to<br>
route to the network of the SDP IP (c=IN IP4 12.34.56.78.)?<br>
<br>
<br>
<br>
BR<br>
<br>
<br>
Max M.<br>
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[mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.<u></u>opensips.org</a>] Im Auftrag von Russ East<br>
Gesendet: Freitag, 13. Januar 2012 20:25<br>
An: <a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a><br>
Betreff: [OpenSIPS-Users] Which way now?<br>
<br>
Hello guys.<br>
<br>
I've been trying to avoid asking for some advice, but three weeks into<br>
studying, watching the webinars, reading the forums, mail-lists, and testing<br>
I've reached a point of "Now What!". I can't find a clear path in the<br>
documentation for what I am trying to do, even though the task seems very<br>
simple!<br>
<br>
I put in a second Cisco 5400 for our local DID numbers (Direct Inward Dial -<br>
not dialog ID) from the PSTN over VoIP. The PSTN will only send to one IP<br>
and will not accept redirects, so I though I just needed a "simple" proxy<br>
and load balance to the ciscos. Seven or eight years ago I helped a friend<br>
setup openSER serving two asterisks - no problem. But I am having problems<br>
here going to the ciscos and am trying to find which combination of modules<br>
I need to use...<br>
<br>
I believe I need to go to topology hiding and a media proxy. Do I use<br>
rtp_proxy or something else like MediaProxy?<br>
<br>
Here's what I want to accomplish... I need to successfully send the calls to<br>
the ciscos, balanced. All are public IP addresses.<br>
<br>
Attached is a sample of my most recent failed attempt with call scenario<br>
data, mysql data, route script and sip trace.<br>
<br>
Any advice or examples would be appreciated!<br>
<br>
<br>
Thank you,<br>
Russ<br>
<br>
<br>
<br>
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