Are you talking about having an SSH tunnel and route your SIP traffic through it !!?
On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming <[email protected]>wrote: > On 01/03/2012 10:03 AM, Patrick Lists wrote: > >> On 03-01-12 16:24, Danny Nicholas wrote: >> >>> Hello List, >>> >>> I work in an environment where I have to request IPTABLES changes rather >>> than doing them myself. Is there a way to utilize my SSH (port 22) to >>> get a functional (and with good sound) Asterisk installation with >>> multiple channels up without requesting the 5060(SIP) 5061 (TLS) and >>> UDP/RTP (usually 10001-20000) IPTABLES allowances? >>> >> >> Not with SIP as it needs a port for signaling (usually 5060) and RTP >> ports for sending the actual voice packets. So for SIP you will always >> need multiple ports. If you can use IAX then you could use port 22 as >> IAX only needs one port. The question is how are you going to SSH into >> the box if you use the SSH port for Asterisk? >> > > It is not practical (although not impossible) to run UDP over an SSH > tunnel. Since VoIP media is generally transported over UDP, this will be a > major obstacle. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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