Krishna Sumanth Chava wrote: > HI Robb, > I had the checked the IP Office and i see that in the SIP Line > Settings an option [checkbox] that says (Use Tel URI), which is > unchecked. But i still get the Tel:+ in the SIP Header (even when it > is turned on or off). > > "you need to make sure the sip dial command in the ipoffice is set to > dial 9n; > feature dial > code n" > > do you mean that i need to program this in the ARS of the avaya IP office? > > i have version 4.1(9) firmware on the Avaya IP small Office. Can you > share me on what Firmware version of avaya IP small Office, you got > the Asterisk and avaya talking to each other. > > Thanks > Krishna > > > > > On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > 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/> > <http://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]> > > <mailto:[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/> <http://10.10.8.2 > <http://10.10.8.2/>> is avaya and 10.10.8.1 <http://10.10.8.1/> > > <http://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 <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 > <http://www.api-digital.com/> -- > > 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 -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users sorry its something like
dial 9n; feature dial code n"@192.168.0.1" where the ip address is the asterisk box 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
