I would like to setup up a remote office with a half dozen or so SIP phones connected to an asterisk server via a WAN link. To conserve bandwidth I would like the phones to be able to re-invite when they call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up re-invites, any experience with that? Assuming no NAT, what should be expected in this setup? I know the transfer option in asterisk would not work, but I do not think that is a big deal since any re-invited calls would be user to user, with little or no need to transfer. As long as the SIP termination peers I am using are set to canreinvite=no then a call between the users and a remote party would not be re-invited, since the peer terminating the call is set to no, correct? Can someone share some experiences wit this type of setup? Are there other real issues to look out for or be aware of? I am really just trying to avoid having another asterisk box in the remote site to maintain, but do not want to waste bandwidth on calls going across the office. Thanks for taking the time to share your wisdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050915/0630a030/attachment.htm
If these phones are all to be in a single location I'd deploy a remote Asterisk box and run an IAX trunk between remote and local sites. That'll save more bandwidth than having a potential 5 individual SIP sessions running over your link. Also, with the addition of an analogue card such as the TDM400 series you'll have survivability should your link go down. If you don't add a phone line to the remote site how will they be able to call 911 etc? Mark Damon Estep wrote:> I would like to setup up a remote office with a half dozen or so SIP > phones connected to an asterisk server via a WAN link. To conserve > bandwidth I would like the phones to be able to re-invite when they call > each other. > > > > The phones will be Polycom, Cisco, or Snom. > > > > I may or may not use NAT. Seems like the NAT would really mess up > re-invites, any experience with that? > > > > Assuming no NAT, what should be expected in this setup? > > > > I know the transfer option in asterisk would not work, but I do not > think that is a big deal since any re-invited calls would be user to > user, with little or no need to transfer. > > > > As long as the SIP termination peers I am using are set to > canreinvite=no then a call between the users and a remote party would > not be re-invited, since the peer terminating the call is set to no, > correct? > > > > Can someone share some experiences wit this type of setup? Are there > other real issues to look out for or be aware of? > > > > I am really just trying to avoid having another asterisk box in the > remote site to maintain, but do not want to waste bandwidth on calls > going across the office. > > > > Thanks for taking the time to share your wisdom. > > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com
> If these phones are all to be in a single location I'd deploy a remote > Asterisk box and run an IAX trunk between remote and local sites. > That'll save more bandwidth than having a potential 5 individual SIP > sessions running over your link.One more potential point of failure, rather not.> > Also, with the addition of an analogue card such as the TDM400 series > you'll have survivability should your link go down.That's what cell phones are for. Each users extensions fails over to an individual cell number in link down situations and just for convenience.> > If you don't add a phone line to the remote site how will they be able > to call 911 etc?We have a 911 solution called PS/ALI that allows us to update the ALI information for each DID (ANI) and we route the 911 calls to a selective router form the main site. Link down - use the cell phone.> > Mark > > Damon Estep wrote: > > I would like to setup up a remote office with a half dozen or so SIP > > phones connected to an asterisk server via a WAN link. To conserve > > bandwidth I would like the phones to be able to re-invite when theycall> > each other. > > > > > > > > The phones will be Polycom, Cisco, or Snom. > > > > > > > > I may or may not use NAT. Seems like the NAT would really mess up > > re-invites, any experience with that? > > > > > > > > Assuming no NAT, what should be expected in this setup? > > > > > > > > I know the transfer option in asterisk would not work, but I do not > > think that is a big deal since any re-invited calls would be user to > > user, with little or no need to transfer. > > > > > > > > As long as the SIP termination peers I am using are set to > > canreinvite=no then a call between the users and a remote partywould> > not be re-invited, since the peer terminating the call is set to no, > > correct? > > > > > > > > Can someone share some experiences wit this type of setup? Are there > > other real issues to look out for or be aware of? > > > > > > > > I am really just trying to avoid having another asterisk box in the > > remote site to maintain, but do not want to waste bandwidth on calls > > going across the office. > > > > > > > > Thanks for taking the time to share your wisdom. > > > > > > > > > > > > > > > > > >------------------------------------------------------------------------> > > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Mark, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users