[asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination
The setup is as follows: SIP phone registers via international link to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 via Zaptel Channels need to be hairpinned from Box 1 to 2. How is sip.conf configured on Box 1 and 2 so that we don't get an error: "Failed to authenticate user" when 1's extensions.conf uses SIP to dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP phone registering at 1 directly to 2 without first passing through 2? Tx Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination
Shaun Wingrin wrote: > The setup is as follows: SIP phone registers via international link > to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 > via Zaptel Channels need to be hairpinned from Box 1 to 2. How is > sip.conf configured on Box 1 and 2 so that we don't get an error: > "Failed to authenticate user" when 1's extensions.conf uses SIP to > dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP > phone registering at 1 directly to 2 without first passing through 2? > > Tx > > Shaun > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users This happens through a sip re-invite, the problem you seem to be having is that box 1 is not authenticated to send calls to box 2. Anthony /"Everything should be as simple as possible, but no simpler" - Albert Einstien/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination
> The setup is as follows: SIP phone registers via international link > to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 > via Zaptel Channels need to be hairpinned from Box 1 to 2. How is > sip.conf configured on Box 1 and 2 so that we don't get an error: > "Failed to authenticate user" when 1's extensions.conf uses SIP to > dial Asterisk Box 2 . How do we ensure that RTP traffic flows from > SIP phone registering at 1 directly to 2 without first passing > through 2? I think if you set up a peer for Box 1 on Box 2, and set insecure=port on those peers, that it will not try to auth calls that are from your other asterisk box. Of course, you'd have to make sure in your diaplan that you restricted access to those calls appropriately. For the RTP, setting canreinvite=yes one peers that you want to be able to send media directly to each other should allow the RTP behavior you are looking for, but keep in mind that if there are any NATs between the phones, things can get messy in a hurry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
