yes I have no control over that. Ok we will figure another way. Thanks
On 25 November 2012 07:10, Duncan Turnbull <[email protected]> wrote: > > > On 25/11/2012, at 1:23 PM, Tiago Geada <[email protected]> wrote: > > linux does sort this out and asterisk listens in both interfaces. however > asterisk connects and tells remote end to send rtp back at the same IP > where sip is going trough... > > remote end does try to send it but gets stopped in a firewall.. thus if > asterisk did present a different IP to recieve RTP in its SIP header, this > would not happen! > > > > I think this is outside of asterisk's natural ability > > You may need a proxy server in between you and the Cisco to achieve this > if you can't change the firewall. > > http://forums.asterisk.org/viewtopic.php?f=1&t=84018 > > Have you tried making the preferred route to these addresses go out eth1, > thus getting the required address? > > Ultimately seems odd the firewall allows access in but not out, guessing > you have no control over that? > > Good luck > > Cheers Duncan > > > On 23 November 2012 19:39, Duncan Turnbull <[email protected]> wrote: > >> >> On 24/11/2012, at 2:19 AM, Tiago Geada <[email protected]> wrote: >> >> Hello Folks, I am looking for a way that makes asterisk tell remote SIP >> party that the IP where they will send RTP is not the same as the one I am >> comunicating via SIP >> >> Can this be done anyhow? >> >> I can try and explain: >> >> We have placed a asterisk box in our partners office. >> >> It has eth0 with IP 172.16.1.10 and eth1: 10.34.18.250 >> >> linux has its routes set so it can comunicate with several networks in >> their offices. >> >> now there is a cisco call manager that we need to communicate with. >> Normally via our IP 172.16.1.10, however seems that this cisco uses some >> sort of 'directmedya=yes' and sets both ends speaking RTP with themselves. >> >> There are some extensions in cisco that have a network 10.134.0.0/16that we >> can only comunicate via eth1 >> >> thus when calling cisco (always via eth0) sometimes we need to say that >> OUR IP to recieve RTP is not 172.16.1.10, but 10.34.18.250 >> >> >> This is a routing issue, not asterisk I think. You are saying you route >> to cisco via eth0, it sets up connections to its end points and then drops >> out of the media flow, but the end points have no route to the eth0 address >> so they fail >> >> Linux usually sorts this out and asterisk replies on the address of the >> interface it sends out with. So for the most part the response in my >> experience if its going out eth1 should use the eth1 ip address. >> >> If you can get to it via eth0 and thats the preferred route then it will >> have the eth0 address. If so why can't you change your routing table to use >> eth1 when you need to go to the cisco then you will have the right address >> and the far extensions can respond to you correctly >> >> Or change the cisco network endpoints so they can successfully access >> your address on eth0 >> >> >> can this be done? >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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