This symptom usually means you are doing an attended transfer instead of a blind transfer.
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland Sent: Monday, January 16, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SayDigits playback doesn't always work Ok, got it. Indeed, starting with Answer() helped. But I still don't understand why the parking feature isn't working then. I used the sample config. Transfer the call to 700, playback of the lot is being executed, but I hear nothing. Probably the same problem, but how do I change this? This is the call that doesn't work. Then when I call 200, I see this: [Jan 16 15:54:29] == Using SIP RTP CoS mark 5 [Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse for Notify User 001565150F04.1 [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] Answer("SIP/000B822FD265-0000003e", "") in new stack [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] BackGround("SIP/000B822FD265-0000003e", "main-menu") in new stack [Jan 16 15:54:29] -- <SIP/000B822FD265-0000003e> Playing 'main-menu.gsm' (language 'nl') [Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] WaitExten("SIP/000B822FD265-0000003e", "5") in new stack [Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-0000003e [Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] Wait("SIP/000B822FD265-0000003e", "2") in new stack [Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] SayDigits("SIP/000B822FD265-0000003e", "123") in new stack [Jan 16 15:54:36] -- <SIP/000B822FD265-0000003e> Playing 'digits/1.gsm' (language 'nl') [Jan 16 15:54:36] -- <SIP/000B822FD265-0000003e> Playing 'digits/2.gsm' (language 'nl') [Jan 16 15:54:37] -- <SIP/000B822FD265-0000003e> Playing 'digits/3.gsm' (language 'nl') [Jan 16 15:54:37] -- Auto fallthrough, channel 'SIP/000B822FD265-0000003e' status is 'UNKNOWN' [Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle for Notify User 001565150F04.1 This call works perfectly. What am I missing? In my sip.conf I have: [stumpel-zwaag](!) ; create template for our devices type=friend ; the channel driver will mathc on username first, IP second context=StumpelZwaag ; this is where calls from the device will enter the dialplan host=dynamic ; the device will register with asterisk ;nat=yes ; assume the device is behind nat secret=xxx ; a secure password for this device dtmfmode=auto ; accept touch-tones from devices, negotiated automatically disallow=all ; reset with voice codecs to accept from, and request to, the device allow=alaw ; which audio codecs we accept from canreinvite=nonat -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
