Jun 15 11:56:10] WARNING[8444]: app_dial.c:1610 dial_exec_full: Invalid timeout specified: 'r'. Setting timeout to infinite From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Gopalakrishnan A.N Sent: Tuesday, June 15, 2010 12:44 AM To: asterisk-ss7 at lists.digium.com Subject: Re: ringback problem Hi, Try the dialplan like this to get Ringback Tone, [wholesale] exten => _473.,1,Dial(DAHDI/g1/${EXTEN},r) exten => _473.,n,Hangup On Mon, Jun 14, 2010 at 10:28 PM, dave george <dgeorge at teletoneinc.com> wrote: One of my customer is not getting any ringback from me. He is sending sip to my asterisk ss7 box using libss7 with TE410P card. I tried the various option (yes, no and never) for progressinband in the sip profile and none worked. Customer is using genband SBC The customer wants: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP This is how we dial the extension [wholesale] exten => _473.,1,Dial(DAHDI/g1/${EXTEN}) exten => _473.,n,Hangup customer profile [customer1] type=peer context=wholesale host=x.x.x.x nat=no canreinvite=no progressinband=yes dtmfmode=rfc2833 insecure=port disallow=all allow=g729 Thanks, Dave George 1 561 674 3838 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -- Thank you with regards, Gopalakrishnan A.N, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100615/c6d699ec/attachment.htm