Re: [asterisk-users] Incoming SIP call is rejected always.
- Original Message - > From: "Yaroslav Panych" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, April 17, 2012 6:56:17 PM > Subject: Re: [asterisk-users] Incoming SIP call is rejected always. > > 2012/4/18 Matthew Jordan : > > I imagine that this is the case, as ASTERISK-19601 noted that > > when this situation occurs, the NOTICE message indicates that > > there is a failure to match the extension, as opposed to a failure > > to match an allowed domain. > > Yes, it was hell to detect real error cause(I was forced to learn how > to debug in KDevelop in less than four hours). Yes, it looks like > ASTERISK-19601. But still I cannot understand why asterisk extracts > wrong domain from request. > > However, in your SIP configuration you have set > > allowexternaldomains to no. > Yes, it is intended. > > > Without knowing the URI the INVITE request was addressed to, its > > difficult to say what might be the actual cause of this. > I first letter I have provided CLI log which contains full request > packets(Authless and authed INVITE included). > > Probably I do not understand how to configure Asterisk: > I have one asterisk. It serves SIP domain example.com. This asterisk > must be able to establish session with registered client of this > account and also must be able to accept incoming sessions. No > sessions > with 3rd-party accounts on 3rd-party domains allowed to established. > How I should setup this asterisk? Well, I can't tell you how to configure your Asterisk server. However, I can tell you why Asterisk rejected the INVITE request. The URI that the INVITE request was addressed to is 4001020@192.168.8.2:5060. The domain portion of this URI is 192.168.8.2. Hence, the allowed domains need to include that particular IPv4 address. Looking at the allowed domains you've specified in sip.conf, we have: domain=sop-korniychuk domain=192.168.8.1 domain=192.168.8.1:5062 So, since the INVITE request does not match any of those three domains, its rejected. Note: I noticed that you have autodomain set to yes; I'm going to assume that the IPv4 address 192.168.8.2 is not associated with the server. Matt > -- > _ > -- 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] Incoming SIP call is rejected always.
2012/4/18 Matthew Jordan : > I imagine that this is the case, as ASTERISK-19601 noted that > when this situation occurs, the NOTICE message indicates that > there is a failure to match the extension, as opposed to a failure > to match an allowed domain. Yes, it was hell to detect real error cause(I was forced to learn how to debug in KDevelop in less than four hours). Yes, it looks like ASTERISK-19601. But still I cannot understand why asterisk extracts wrong domain from request. > However, in your SIP configuration you have set allowexternaldomains to no. Yes, it is intended. > Without knowing the URI the INVITE request was addressed to, its > difficult to say what might be the actual cause of this. I first letter I have provided CLI log which contains full request packets(Authless and authed INVITE included). Probably I do not understand how to configure Asterisk: I have one asterisk. It serves SIP domain example.com. This asterisk must be able to establish session with registered client of this account and also must be able to accept incoming sessions. No sessions with 3rd-party accounts on 3rd-party domains allowed to established. How I should setup this asterisk? -- _ -- 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] Incoming SIP call is rejected always.
Without knowing the URI the INVITE request was addressed to, its difficult to say what might be the actual cause of this. However, in your SIP configuration you have set allowexternaldomains to no. That implies that if the domain of the URI does not match any of the allowed domains you have set, that the INVITE request will be rejected. I imagine that this is the case, as ASTERISK-19601 noted that when this situation occurs, the NOTICE message indicates that there is a failure to match the extension, as opposed to a failure to match an allowed domain. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org - Original Message - > From: "Yaroslav Panych" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, April 17, 2012 4:58:14 PM > Subject: Re: [asterisk-users] Incoming SIP call is rejected always. > > 2012/4/17 Danny Nicholas : > > Maybe it needs to be _4001020? > > > > Not, it doesn't. Actually I have traced this incoming call step by > step. Real reason it refuses - wrong domain. But why it wrong - have > not any idea. > > -- > _ > -- 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] Incoming SIP call is rejected always.
2012/4/17 Danny Nicholas : > Maybe it needs to be _4001020? > Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain. But why it wrong - have not any idea. -- _ -- 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] Incoming SIP call is rejected always.
Maybe it needs to be _4001020? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav Panych Sent: Tuesday, April 17, 2012 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Incoming SIP call is rejected always. Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- 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] Incoming SIP call is rejected always.
Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- 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
