Re: [asterisk-users] Question about SS7
On Wed, 14 May 2008 17:06:54 -0400, Alexander Lopez wrote: SS7 helps carriers maximize the use of the circuits that interconnect them with others. Instead of using a channel and having it open for 30 seconds as the call is setup, user gets signaling (busy, ringing, not in service), and call is torn down. It can get the result in a split second with out using any of my channels, all out of band and digital rather than analog, (see 2600 signaling) simplistically, ss7 is like sip which sets up the call, and the circuit itself is the rtp streams which are then built when the call is connected. likewise, you can have the sip exchange go through one path/route and rtp through another. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SS7
Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of implementing Asterisk, what kind of SS7 support is needed? Is SS7 something needs to be concerned about when using Asterisk with T1/E1? I hope someone can help me to clearify these doubts that I am having. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SS7
SS7 does NOT play a roll in VoIP. The SS7 signaling that you are describing is not really SS7 but signaling over a PRI using ISDN that your provider uses to exchange information via SS7 to the other carriers. To be blunt and I do not mean to be condescending in any way, but, if you are using Asterisk and do not know what SS7 is, you don't need to worry about it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mark morreny Sent: Wednesday, May 14, 2008 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about SS7 Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of implementing Asterisk, what kind of SS7 support is needed? Is SS7 something needs to be concerned about when using Asterisk with T1/E1? I hope someone can help me to clearify these doubts that I am having. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
