<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">Doug<br />
<br />
Send me step by step directions (off list) on how to reproduce the issue (dialplan code example is best) you are having and I will open and ticket with my Grandstream contacts. I will push them for a fix for this bug ASAP as we use these phones very aggressively. I have to be able to reproduce it in our test lab and We can usually get it into the fix cycle quickly. The guys at Grandstream are good about fixing stuff if you server it up on a platter for them. Please include your full contact info so I can correspond with you if necessary.<br />
<div>Thanks<br />
<br />
Bryant Zimmerman (ZK Tech Inc.)<br />
616-855-1030 Ext. 2003</div>
<br />
<br />
<span style="font-family: tahoma,arial,sans-serif; font-size: 10pt;"><hr width="100%" size="2" align="center">
</hr><b>From</b>: "Doug Crompton" <doug@crompton.com><br />
<b>Sent</b>: Saturday, April 21, 2012 10:40 AM<br />
<b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-users] GXP1400</span><br />
<br />
Just a note to say that updating the phone to the latest beta code -<br />
1.0.3.30 - mostly solved the problems I was having. It now gives ring back<br />
tones when making a local call however the early dial feature still does<br />
not work as it should and did in earlier (GXP280) phones. When making a<br />
call and hitting the digits the invite to Asterisk should be sent at the<br />
release of the key. In the GXP1400 it is sent about 100ms after the key is<br />
hit. Thus when you select a key you hear a short 100ms or so of the tone<br />
and the invite is sent to Asterisk even though you have not released the<br />
key. It does not sound natural and is not the way humans are use to<br />
interacting with a phone. I am hoping that Grandstream corrects this on a<br />
future firmware release.<br />
<br />
<br />
On Thu, 12 Apr 2012, Doug Crompton wrote:<br />
<br />
> Some more input on this. I tried adding an explicit 'r' option to the dial<br />
> command with no change. I also tried the 'progressinband=yes' setting in<br />
> the section for this phone in sip.conf. No difference. No call tones and<br />
> no or little blip DTMF tones in the handset when making a call. Offsite<br />
> calls return call "ring" tone.<br />
><br />
> Doug<br />
><br />
><br />
> On Thu, 12 Apr 2012, Doug Crompton wrote:<br />
><br />
> > It is Asterisk 1.2 - yes I know old but it works for my application. The<br />
> > problem would not be gain, I have plenty of handset gain. It is the fact<br />
> > that the ring confirmation tones on local calls, which would be generated<br />
> > by asterisk are not there. They work fine on my two other older<br />
> > Grandstream phones. Also when I enter digits I do not hear the DTMF tones<br />
> > or only a very short blip. It completes calls fine.<br />
> ><br />
> > I have tried with and without 'early call' and bot 'in band' and 'RFC2833'<br />
> > DTMF. No change with either.<br />
> ><br />
> > Any ideas?<br />
> ><br />
> > Doug<br />
> ><br />
> ><br />
> ><br />
> > On Thu, 12 Apr 2012, Danny Nicholas wrote:<br />
> ><br />
> > > You don't state which version of Asterisk you have this hooked up to, but<br />
> > > dependent on that you might have rxgain and txgain available to you.<br />
> > ><br />
> > > -----Original Message-----<br />
> > > From: asterisk-users-bounces@lists.digium.com<br />
> > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Doug Crompton<br />
> > > Sent: Thursday, April 12, 2012 3:36 PM<br />
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion<br />
> > > Subject: [asterisk-users] GXP1400<br />
> > ><br />
> > > Just installed a Grandstream GXP1400 - I have several other older<br />
> > > Grandstream phones installed. This one for some reason is operating<br />
> > > differently. It works but I get no audible ring confirmation in the headset<br />
> > > when I make a local call and I also only hear a blip of the DTMF ddigits or<br />
> > > nothing at all in the headset as I key them in. Everything works OK, calls<br />
> > > etc. just the audible feedback. Using an old Asterisk but all the other<br />
> > > phones have worked fine. Is there some setting in Asterisk or the<br />
> > > Grandstream I should be looking at to correct this?<br />
> > ><br />
> > > Doug<br />
> > ><br />
> > ><br />
> > > --<br />
> > > _____________________________________________________________________<br />
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to<br />
> > > Asterisk? Join us for a live introductory webinar every Thurs:<br />
> > > http://www.asterisk.org/hello<br />
> > ><br />
> > > asterisk-users mailing list<br />
> > > To UNSUBSCRIBE or update options visit:<br />
> > > http://lists.digium.com/mailman/listinfo/asterisk-users<br />
> > ><br />
> > ><br />
> > > --<br />
> > > _____________________________________________________________________<br />
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br />
> > > New to Asterisk? Join us for a live introductory webinar every Thurs:<br />
> > > http://www.asterisk.org/hello<br />
> > ><br />
> > > asterisk-users mailing list<br />
> > > To UNSUBSCRIBE or update options visit:<br />
> > > http://lists.digium.com/mailman/listinfo/asterisk-users<br />
> > ><br />
> ><br />
> ><br />
> ><br />
> > "Although it is not true that all conservatives are stupid people,<br />
> > it is true that most stupid people are conservative"<br />
> ><br />
> > ****************************<br />
> > * Doug Crompton *<br />
> > * Richboro, PA 18954 *<br />
> > * 215-355-5307 *<br />
> > * *<br />
> > * doug@crompton.com *<br />
> > * http://www.crompton.com *<br />
> > ****************************<br />
> ><br />
> ><br />
> ><br />
> > --<br />
> > _____________________________________________________________________<br />
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br />
> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br />
> > http://www.asterisk.org/hello<br />
> ><br />
> > asterisk-users mailing list<br />
> > To UNSUBSCRIBE or update options visit:<br />
> > http://lists.digium.com/mailman/listinfo/asterisk-users<br />
> ><br />
><br />
><br />
><br />
> "Although it is not true that all conservatives are stupid people,<br />
> it is true that most stupid people are conservative"<br />
><br />
> ****************************<br />
> * Doug Crompton *<br />
> * Richboro, PA 18954 *<br />
> * 215-355-5307 *<br />
> * *<br />
> * doug@crompton.com *<br />
> * http://www.crompton.com *<br />
> ****************************<br />
><br />
><br />
><br />
> --<br />
> _____________________________________________________________________<br />
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br />
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br />
> http://www.asterisk.org/hello<br />
><br />
> asterisk-users mailing list<br />
> To UNSUBSCRIBE or update options visit:<br />
> http://lists.digium.com/mailman/listinfo/asterisk-users<br />
><br />
<br />
<br />
<br />
"Although it is not true that all conservatives are stupid people,<br />
it is true that most stupid people are conservative"<br />
<br />
****************************<br />
* Doug Crompton *<br />
* Richboro, PA 18954 *<br />
* 215-355-5307 *<br />
* *<br />
* doug@crompton.com *<br />
* http://www.crompton.com *<br />
****************************<br />
<br />
<br />
<br />
--<br />
_____________________________________________________________________<br />
-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br />
New to Asterisk? Join us for a live introductory webinar every Thurs:<br />
http://www.asterisk.org/hello<br />
<br />
asterisk-users mailing list<br />
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<br /></span>