Krishna Sumanth Chava wrote: > Hi * Users, > > I ran into a problem when I was trying to communicate an avaya IP > Office talk to asterisk with SIP Trunking. I had successful calls from > asterisk to Avaya but not from avaya to asterisk. > > Can someone provide me insight on how to address it or the path to > resolve it. > > The error I get is mentioned below: (dialing 32564 from avaya to asterisk) > > "[Nov 6 17:14:23] WARNING[6227]: chan_sip.c:8686 get_destination: > Huh? Not a SIP header (Tel:+32564)? > [Nov 6 17:14:23] NOTICE[6227]: chan_sip.c:13774 > handle_request_invite: Call from 'avayanew' to extension 'Tel:+32564' > rejected because extension not found." > > A SIP Debug of the packet when this happens on asterisk CLI is > > "<--- SIP read from 10.10.8.2:5060 <http://10.10.8.2:5060> ---> > ACK Tel:+32564 SIP/2.0 > Via: SIP/2.0/UDP > 10.10.8.2:5060;rport;branch=z9hG4bKb8f50a43f8fce87fda53573e96e498a9 > From: avayanew <sip:[EMAIL PROTECTED]>;tag=d60c0430c7b26cbd > To: Tel:+32564;tag=as51355066 > Call-ID: [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > CSeq: 152795667 ACK > Max-Forwards: 70 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO > Content-Length: 0" > > Note: 10.10.8.2 <http://10.10.8.2> is avaya and 10.10.8.1 > <http://10.10.8.1> is asterisk > > As I understand, we are getting a Tel URI and a "+" like in e.164 > format and then the number dialed (32564)from avaya. These errors are > coming on asterisk console when I try to dial a call from Avaya IP > Phone over its SIP trunk on to the asterisk. We probably have to strip > the 'Tel:+', so that the asterisk gets the number and thus follows the > dialplan programmed in extensions file. > > Please advise. Any help is appreciated. > > Thanks as always > > Regards > Krishna > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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 you need to make sure the sip dial command in the ipoffice is set to dial 9n; feature dial code n
in system the set the dial delay timer to 4 seconds and the dial delay count to 1 this will allow 4 seconds in between each digit there is a setting on the ipo to change the TEL:+ setting to url setting cannot remember wher it is but it in the sip trunk settings robb _______________________________________________ -- 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