Re: [asterisk-users] Realtime peers and trunks coming from the same IP
Thanks Sammy, I think I'll stop using SIP realtime. Regards, Ricardo. On Mon, May 21, 2012 at 5:14 AM, SamyGo wrote: > Hello Ricardo, > The reason why your asterisk refused the calls from phone registering on > SIP proxy is that it only gets INVITE of the call from: a user that is > defined BUT Not Registered within asterisk. > The easy way of solving this is > 1- Stop asterisk SIP realtime and let only the SIP proxy handle > registrations. > 2- Tell asterisk to accept calls from the SIP proxy only (create a SIP > peer for proxy) > This will make everything work. > > Regards, > Sammy. > > On Sat, May 19, 2012 at 9:15 PM, Ricardo Carvalho < > rjcarvalho.li...@gmail.com> wrote: > >> I use an SBC to protect my pool of asterisk servers and as trunking >> endpoint with SIP Telcos. Now I'm trying to implement SIP phone >> registration to be delegated through the SBC, as a proxy. >> >> It doesn't work. It just works when I don't use realtime peers at the >> asterisk servers. Using realtime SIP peers, since there is one SIP phone >> that gets his registration delegated through the SBC, any inbound call that >> tries to reach any asterisk server, coming from any SIP Telco trunk ended >> at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC >> as the IP of the phone that has been registered, it "thinks" that those >> calls coming from the SBC are calls coming from that phone, and it refuses >> them with "401 Unauthorized" replies. I'm using asterisk 1.8.11. >> >> How can I surpass this problem? Is there any configuration that I'm >> lacking on, or is this a limitation of asterisk? >> >> Thanks, >> Ricardo. >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime peers and trunks coming from the same IP
Hi Sammy go Can you help me with my problem I have asterisk 1.8 and i am using asterisk-gui 2.0, and in asterisk-gui 2.0 the voice prompt menu which is used for custom voice recording for IVR is not working and not recording. Can u tell me how to defualt this feature. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime peers and trunks coming from the same IP
Hello Ricardo, The reason why your asterisk refused the calls from phone registering on SIP proxy is that it only gets INVITE of the call from: a user that is defined BUT Not Registered within asterisk. The easy way of solving this is 1- Stop asterisk SIP realtime and let only the SIP proxy handle registrations. 2- Tell asterisk to accept calls from the SIP proxy only (create a SIP peer for proxy) This will make everything work. Regards, Sammy. On Sat, May 19, 2012 at 9:15 PM, Ricardo Carvalho < rjcarvalho.li...@gmail.com> wrote: > I use an SBC to protect my pool of asterisk servers and as trunking > endpoint with SIP Telcos. Now I'm trying to implement SIP phone > registration to be delegated through the SBC, as a proxy. > > It doesn't work. It just works when I don't use realtime peers at the > asterisk servers. Using realtime SIP peers, since there is one SIP phone > that gets his registration delegated through the SBC, any inbound call that > tries to reach any asterisk server, coming from any SIP Telco trunk ended > at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC > as the IP of the phone that has been registered, it "thinks" that those > calls coming from the SBC are calls coming from that phone, and it refuses > them with "401 Unauthorized" replies. I'm using asterisk 1.8.11. > > How can I surpass this problem? Is there any configuration that I'm > lacking on, or is this a limitation of asterisk? > > Thanks, > Ricardo. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime peers and trunks coming from the same IP
I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now I'm trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesn't work. It just works when I don't use realtime peers at the asterisk servers. Using realtime SIP peers, since there is one SIP phone that gets his registration delegated through the SBC, any inbound call that tries to reach any asterisk server, coming from any SIP Telco trunk ended at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC as the IP of the phone that has been registered, it "thinks" that those calls coming from the SBC are calls coming from that phone, and it refuses them with "401 Unauthorized" replies. I'm using asterisk 1.8.11. How can I surpass this problem? Is there any configuration that I'm lacking on, or is this a limitation of asterisk? Thanks, Ricardo. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
