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/UDP 12.34.56.78:5060. From: <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, 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, <[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].**org<[email protected]> >> [mailto:users-bounces@lists.**opensips.org<[email protected]>] >> Im Auftrag von Russ East >> Gesendet: Freitag, 13. Januar 2012 20:25 >> An: [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 >> >> >> >> ----- >> eMail ist virenfrei. >> Von AVG überprüft - www.avg.de >> Version: 10.0.1416 / Virendatenbank: 2109/4140 - Ausgabedatum: 13.01.2012 >> >> >> ______________________________**_________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >> >> > ______________________________**_________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >
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