Re: [Asterisk-Users] conditional canreinvite
Hi, Thanks for that cool info. It will help me in the days to come Great going... Dan On 25/01/06, hugolivude <[EMAIL PROTECTED]> wrote: Guys,Have a look for my posting:"How to keep Asterisk (1.2) out of the media path" A gentleman named Tony Jago provided some awesome info. I've posted it below, but you might want to look at my posting for context:1) Could someone confirm that I'll need to have canreinvite=yes insip.conf for both the Xlite and the Polycoms in order to bypass * fromthe media path?This is correct.2) Does the Polycom & XLite support reinvite? I believe so.3) Does reinvite work if you're behind a nat? i.e. if I have nat=yes,does this mean I _have_ to have canreinvite=no?No. You need to have the nat set up correctly. This means you need to put port forwarding rules in for each and every phone for it's sip and rtpports. This means you will have to reconfigure each phone to use a different port. eg.phone 10.0.0.1. SIP port 5060 and RTP ports 8001-8010.phone 10.0.0.2 SIP port 5061 and RTP ports 8011-8020. phone 10.0.0.3. SIP port 5062 and RTP ports 8021-8030.etc etc. on your firewall, you need to map incoming ports 5060 -> 10.0.0.1 and 8001-8010 -> 10.0.0.15061 -> 10.0.0.2 and 8011-8020 -> 10.0.0.2 etc etc. You need to turn on NAT support on each phone. What you are doing here is allowing each and every phone to work in its own right across the NAT gateway. After you have finished. Each and every phone should be able to make and receive calls from anywhere on the internet (without going through Asterisk). At this point, if you sacrifice a few chickens and a walrus you may get itall to work.Finally: I have a suspicion that using a NAT router will prevent me from eliminating Asterisk from the media path. I am currently running aLinksys WRT54G with Talisman to get QOS. Any recommendations for analternate QOS router? Ideally it will also support multiple sub-domains... You can do all sorts of stuff with your WRT54G. Running openser on yourWRT54G could in theory do what your looking for. There are plenty of WRT54G firmwares that let you do nifty VoIP things. You can even install asteriskon your WRT54G. Check out www.openwrt.org Hope this is some help. PS: I found a bug in asterisk's re-invite code that in some cases makes asterisk push out an invalid SIP packet. If you see anything like this, letme know and I can send you the patch. On 1/24/06, David Thomas <[EMAIL PROTECTED] > wrote:> That is the way way SER works. I too am very interested to know if> this can be done with Asterisk.> > David> > On 1/12/06, Pavel Jezek < [EMAIL PROTECTED]> wrote:> > Hi, I have asterisk on public IP and phones in two locations behind> > firewall/nat, > > - when I have nat=yes and canreinvite=no, this is working fine, but rtp > > stream must go _always_ through asterisk, even if phones talk inside> > their locations> > - when I have nat=yes and canreinvite=yes, phones can speak only inside > > their location and rtp stream is connected directly between phones (this > > is, imho, correct and logical), but,> > is possible to combine both, so do reinvite only "within" e.g . one> > context and disable reinvite when connecting phones between two context, > > or any better option exist/planned how to solve?> > thanks> > PJ> > ___ > > --Bandwidth and Colocation provided by Easynews.com --> >> > Asterisk-Users mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-users> >> ___> --Bandwidth and Colocation provided by Easynews.com --> > Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users > ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conditional canreinvite
Guys,Have a look for my posting:"How to keep Asterisk (1.2) out of the media path"A gentleman named Tony Jago provided some awesome info. I've posted it below, but you might want to look at my posting for context:1) Could someone confirm that I'll need to have canreinvite=yes insip.conf for both the Xlite and the Polycoms in order to bypass * from the media path?This is correct.2) Does the Polycom & XLite support reinvite?I believe so.3) Does reinvite work if you're behind a nat? i.e. if I have nat=yes,does this mean I _have_ to have canreinvite=no?No. You need to have the nat set up correctly. This means you need to put port forwarding rules in for each and every phone for it's sip and rtpports. This means you will have to reconfigure each phone to use a different port. eg.phone 10.0.0.1. SIP port 5060 and RTP ports 8001-8010.phone 10.0.0.2 SIP port 5061 and RTP ports 8011-8020. phone 10.0.0.3. SIP port 5062 and RTP ports 8021-8030. etc etc.on your firewall, you need to map incoming ports 5060 -> 10.0.0.1 and 8001-8010 -> 10.0.0.15061 -> 10.0.0.2 and 8011-8020 -> 10.0.0.2 etc etc.You need to turn on NAT support on each phone. What you are doing here is allowing each and every phone to work in its own right across the NAT gateway. After you have finished. Each and every phoneshould be able to make and receive calls from anywhere on the internet (without going through Asterisk). At this point, if you sacrifice a few chickens and a walrus you may get itall to work.Finally:I have a suspicion that using a NAT router will prevent me from eliminating Asterisk from the media path. I am currently running aLinksys WRT54G with Talisman to get QOS. Any recommendations for analternate QOS router? Ideally it will also support multiplesub-domains... You can do all sorts of stuff with your WRT54G. Running openser on yourWRT54G could in theory do what your looking for. There are plenty of WRT54G firmwares that let you do nifty VoIP things. You can even install asterisk on your WRT54G. Check out www.openwrt.orgHope this is some help. PS: I found a bug in asterisk's re-invite code that in some cases makes asterisk push out an invalid SIP packet. If you see anything like this, letme know and I can send you the patch. On 1/24/06, David Thomas <[EMAIL PROTECTED]> wrote:> That is the way way SER works. I too am very interested to know if> this can be done with Asterisk.> > David> > On 1/12/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:> > Hi, I have asterisk on public IP and phones in two locations behind> > firewall/nat, > > - when I have nat=yes and canreinvite=no, this is working fine, but rtp> > stream must go _always_ through asterisk, even if phones talk inside> > their locations> > - when I have nat=yes and canreinvite=yes, phones can speak only inside > > their location and rtp stream is connected directly between phones (this> > is, imho, correct and logical), but,> > is possible to combine both, so do reinvite only "within" e.g . one> > context and disable reinvite when connecting phones between two context,> > or any better option exist/planned how to solve?> > thanks> > PJ> > ___ > > --Bandwidth and Colocation provided by Easynews.com --> >> > Asterisk-Users mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-users> >> ___> --Bandwidth and Colocation provided by Easynews.com --> > Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conditional canreinvite
That is the way way SER works. I too am very interested to know if this can be done with Asterisk. David On 1/12/06, Pavel Jezek <[EMAIL PROTECTED]> wrote: > Hi, I have asterisk on public IP and phones in two locations behind > firewall/nat, > - when I have nat=yes and canreinvite=no, this is working fine, but rtp > stream must go _always_ through asterisk, even if phones talk inside > their locations > - when I have nat=yes and canreinvite=yes, phones can speak only inside > their location and rtp stream is connected directly between phones (this > is, imho, correct and logical), but, > is possible to combine both, so do reinvite only "within" e.g. one > context and disable reinvite when connecting phones between two context, > or any better option exist/planned how to solve? > thanks > PJ > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conditional canreinvite
Great posting. I'm keen to learn the answer as well. Hugh -- Forwarded message -- From: Pavel Jezek <[EMAIL PROTECTED]> Date: Jan 12, 2006 9:33 AM Subject: [Asterisk-Users] conditional canreinvite To: asterisk-users@lists.digium.com Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho, correct and logical), but, is possible to combine both, so do reinvite only "within" e.g. one context and disable reinvite when connecting phones between two context, or any better option exist/planned how to solve? thanks PJ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] conditional canreinvite
Hi, I have asterisk on public IP and phones in two locations behind firewall/nat, - when I have nat=yes and canreinvite=no, this is working fine, but rtp stream must go _always_ through asterisk, even if phones talk inside their locations - when I have nat=yes and canreinvite=yes, phones can speak only inside their location and rtp stream is connected directly between phones (this is, imho, correct and logical), but, is possible to combine both, so do reinvite only "within" e.g. one context and disable reinvite when connecting phones between two context, or any better option exist/planned how to solve? thanks PJ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
