The strange thing is its only sometimes my dial string is as follows exten => s,1, Dial (SIP/200,, tT) For that particular route but obviously s,3 as have Ringing () first etc. After she pushes ## 6 times it will go thru sometimes. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Kevin Larsen <kevin.larsen at pioneerballoon.com> </div><div>Date:16/02/2015 17:11 (GMT+02:00) </div><div>To: Andrew Colin <andrew at convergedgroup.net>,Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> </div><div>Subject: Re: [asterisk-users] BlindXfer Sensitivity </div><div> </div>> Hi Guys> > We have a client running on a polycom vvx400 IP phone on our > asterisk 1.8.18 system > > The issue we have is the switchboard lady uses ## to transfer calls > but sometimes it just does not work and just plays the DTMF tone to > the calling party. > > Is there any way to adjust the sensitivity of the blindxfer feature? > > The polycom Transfer button is useless as there is a big delay > until it apprears > > I would greatly appreciate any adviceIt seems weird that this would be some kind of sensitivity to the DTMF tones. The first thing I would look for is on a call that she cannot blind transfer, check how the Dial command was used to reach her. Does it have the proper use of the tT options (depending on whether she called them or they called her)? I would almost bet there is a call path that occurs which doesn't have the proper options set to allow the transfer. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150216/53741f0d/attachment.html>
On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin <andrew at vsave.co.za> wrote:> The strange thing is its only sometimes my dial string is as follows > > exten => s,1, Dial (SIP/200,, tT) > > For that particular route but obviously s,3 as have Ringing () first etc. > After she pushes ## 6 times it will go thru sometimes. > >How is the DTMF being transmitted from the phone to Asterisk? RFC2833, in-band, SIP INFO...? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150216/07c168e4/attachment.html>
On 16/2/15 4:13 pm, Andrew Colin wrote:> The strange thing is its only sometimes my dial string is as follows > exten => s,1, Dial (SIP/200,, tT) > For that particular route but obviously s,3 as have Ringing () first etc. > After she pushes ## 6 times it will go thru sometimes.Are you sure it's a DTMF sensitivity problem rather than a delay problem? I've had several sites where the default DTMF timeout of 0.5 seconds is too short for users to achieve, and have set featuredigittimeout (in features.conf) to 3 seconds to give them more time to press the combinations they need to press. Kind regards, Chris -- This email is made from 100% recycled electrons
RFC2833 The strange thing is how asterisk is not registering she has pushed ## on those "Rare" occiasions"> On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin <andrew at vsave.co.za> wrote: > >> The strange thing is its only sometimes my dial string is as follows >> >> exten => s,1, Dial (SIP/200,, tT) >> >> For that particular route but obviously s,3 as have Ringing () first >> etc. >> After she pushes ## 6 times it will go thru sometimes. >> >> > How is the DTMF being transmitted from the phone to Asterisk? RFC2833, > in-band, SIP INFO...? > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > -- > _____________________________________________________________________ > -- 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