You can try changing the priority of '1104' => 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]
tp this '1104' => 2. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] On Tue, Mar 15, 2011 at 6:59 PM, Jerry Geis <[email protected]> wrote: > I am using asterisk 1.8.3. > > I am getting this error: > [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: > Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because > extension not found in context 'smvoice-mediaport'. > > "dialplan show" gives me that the context is present: > > [ Context 'smvoice-mediaport' created by 'pbx_config' ] > '1104' => 1. Goto(smvoice-mediaport-public-address,s,1) > [pbx_config] > 'mediaport_direct' => 1. Goto(smvoice-mediaport-public-address,s,1) > [pbx_config] > 'public_address' => 1. Goto(smvoice-mediaport-public-address,s,1) > [pbx_config] > > > the server is showing the call to 1104 which is valid : > -- Executing [1104@smvoice-sip:1] Dial("SIP/528-00000124", > "SIP/mndemo_to_vizioconfrm104/1104") in new stack > == Using SIP RTP CoS mark 5 > > > Why is my call not going through? > > Jerry > > > > -- > _____________________________________________________________________ > -- 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 > -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: [email protected] W: www.axvoice.com
-- _____________________________________________________________________ -- 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
