Hi Guys, Thanks that did help to resolve my issue. i tried the ."@10.10.8.1" and it worked and i had a successful call but i have the following 2 concerns.
1. We have voice communication from avaya to asterisk now but avaya is forcing asterisk to use only codec G723. if i disable G723, it says no compatible codecs. While the calls from asterisk to avaya are being accepted as "alaw" 2. I am having issues with DTMF. DTMF is not being recognized or being sent from avaya to asterisk. I had connected an Analog phone to the POTS line of the IP Office for this experiment. Also i am having hard time for detecting Hangups. Please advise. Any help is appreciated as i am new to avaya IP office and am much familiar with asterisk. Regards Krishna On Sat, Nov 8, 2008 at 12:28 PM, Robert Boardman <[EMAIL PROTECTED]>wrote: > 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 >
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