In your sip.conf, in trunk parameters use: dtmfmode = INFO Date: Wed, 16 Feb 2011 23:07:16 +0800 From: [email protected] To: [email protected] Subject: Re: [asterisk-users] DTMF not detected, time out
It is somehow back to normal. Nothing change. May be the sip provider problem. However, it lasts for quite a while. Thanks On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif <[email protected]> wrote: You can also append add dtmf logging to cosole and see if dtmf is coming from carrier. From: [email protected] [mailto:[email protected]] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF not detected, time out In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <[email protected]> wrote:Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. From: [email protected] [mailto:[email protected]] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
