Bill Gibbs
2006-Sep-18 14:19 UTC
[asterisk-users] sip.conf for talking to other Asterisk machines
Just curious how most of you are defining SIP peers in sip.conf - for Asterisk boxes talking to each other. Are most of you just making a type=friend connection and a single context or are you separating them out to in/out definitions and contexts? In other words Where voicegw1 is the Asterisk box with the TDM cards for talking to the PSTN, it will receive calls from the PSTN and forward to the appropriate Asterisk box as well as receive calls from the other Asterisk boxes to forward out to the PSTN. Do you on the Asterisk box that contains all the SIP phones define (ie the client to the PSTN Asterisk box and voicegw1 is the one with the PSTN connection) [voicegw1-in] type=user username=virtualpbx1-in secret=1234 host=192.168.1.99 context=voicegw1-in canreinvite=no nat=no qualify=yes allow=all [voicegw1-out] type=peer username=virtualpbx1-out secret=1234 host=192.168.1.99 context=voicegw1-out canreinvite=no nat=no qualify=yes allow=all or [voicegw1] Type=friend Blah Context=voicegw1 And use a single context for inbound/outbound routing? The same would apply to the PSTN Asterisk server. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060918/f471c805/attachment.htm
Forrest Beck
2006-Sep-18 18:51 UTC
[asterisk-users] sip.conf for talking to other Asterisk machines
I use two user's per host one for user and the other peer. Sort of like attahed. I also prefer IAX for communication between asterisk boxes. IAX use's less bandwidth than SIP and it's trunks are alot smaller. If you look at SIP traffic, 80% of it is headers. The headers look just like smtp headers. Even if your clients are using SIP to communicate to asterisk using SIP, the asterisk servers will maintain the trunked connection route the traffic for your SIP phones. On 9/18/06, Bill Gibbs <bgibbs@edurotech.com> wrote:> > > > > Just curious how most of you are defining SIP peers in sip.conf ? for > Asterisk boxes talking to each other. Are most of you just making a > type=friend connection and a single context or are you separating them out > to in/out definitions and contexts? > > > > In other words > > Where voicegw1 is the Asterisk box with the TDM cards for talking to the > PSTN, it will receive calls from the PSTN and forward to the appropriate > Asterisk box as well as receive calls from the other Asterisk boxes to > forward out to the PSTN. > > > > Do you on the Asterisk box that contains all the SIP phones define (ie the > client to the PSTN Asterisk box and voicegw1 is the one with the PSTN > connection) > > [voicegw1-in] > > type=user > > username=virtualpbx1-in > > secret=1234 > > host=192.168.1.99 > > context=voicegw1-in > > canreinvite=no > > nat=no > > qualify=yes > > allow=all > > > > [voicegw1-out] > > type=peer > > username=virtualpbx1-out > > secret=1234 > > host=192.168.1.99 > > context=voicegw1-out > > canreinvite=no > > nat=no > > qualify=yes > > allow=all > > > > or > > > > [voicegw1] > > Type=friend > > Blah > > Context=voicegw1 > > > > And use a single context for inbound/outbound routing? > > > > The same would apply to the PSTN Asterisk server. > > > > > > Bill > _______________________________________________ > --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 > > >-------------- next part -------------- A non-text attachment was scrubbed... Name: 662006_23640_0.png Type: image/png Size: 5116 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060918/3f2fabe9/662006_23640_0.png
Douglas Garstang
2006-Sep-18 20:42 UTC
[asterisk-users] sip.conf for talking to other Asterisk machines
IAX has some pretty severe limitations when it comes to trunking calls between Asterisk boxes. It can't pass variables for example, and any calls to SIP phones at the far end will be treated as IAX calls, which is just nuts. This means you lose a lot of SIP features, like transferring and forwarding. We had to drop IAX and go back to SIP, which is pretty ironic considering IAX stands for Inter Asterisk Exchange. -----Original Message----- From: Forrest Beck [mailto:jonforrest.beck@gmail.com] Sent: Mon 9/18/2006 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] sip.conf for talking to other Asterisk machines I use two user's per host one for user and the other peer. Sort of like attahed. I also prefer IAX for communication between asterisk boxes. IAX use's less bandwidth than SIP and it's trunks are alot smaller. If you look at SIP traffic, 80% of it is headers. The headers look just like smtp headers. Even if your clients are using SIP to communicate to asterisk using SIP, the asterisk servers will maintain the trunked connection route the traffic for your SIP phones. On 9/18/06, Bill Gibbs <bgibbs@edurotech.com> wrote: > > > > > Just curious how most of you are defining SIP peers in sip.conf ? for > Asterisk boxes talking to each other. Are most of you just making a > type=friend connection and a single context or are you separating them out > to in/out definitions and contexts? > > > > In other words > > Where voicegw1 is the Asterisk box with the TDM cards for talking to the > PSTN, it will receive calls from the PSTN and forward to the appropriate > Asterisk box as well as receive calls from the other Asterisk boxes to > forward out to the PSTN. > > > > Do you on the Asterisk box that contains all the SIP phones define (ie the > client to the PSTN Asterisk box and voicegw1 is the one with the PSTN > connection) > > [voicegw1-in] > > type=user > > username=virtualpbx1-in > > secret=1234 > > host=192.168.1.99 > > context=voicegw1-in > > canreinvite=no > > nat=no > > qualify=yes > > allow=all > > > > [voicegw1-out] > > type=peer > > username=virtualpbx1-out > > secret=1234 > > host=192.168.1.99 > > context=voicegw1-out > > canreinvite=no > > nat=no > > qualify=yes > > allow=all > > > > or > > > > [voicegw1] > > Type=friend > > Blah > > Context=voicegw1 > > > > And use a single context for inbound/outbound routing? > > > > The same would apply to the PSTN Asterisk server. > > > > > > Bill > _______________________________________________ > --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 > > >