Re: [Asterisk-Users] Survey: Grandstream improvements.........
Michael T Farnworth wrote: On Fri, 24 Oct 2003, Michael Koehler wrote: :) imo it is called Budge Tone .. "budge" from "move" What will you guys think what the name "HandyTone"imply, which could suggest convenience ? Anything that gets rid of 'Budget' in the name is a good idea in my mind, as long as it isn't changed to CheapTone :-) 'Handy' is quite a good choice I think. If you tried to turn it into SuperTone it might be just viewed as being over the top and unbelievable, so you probably do need something in the middle. You could be right. "DeskTone" would be the right choise then maybe.. Michael
RE: [Asterisk-Users] Survey: Grandstream improvements.........
Why not just the Grandstream 100, 101 102 ? Grand as in Grandiose, Great etc. Stream as in that is what we are doing with the data. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael Koehler Sent: 24 October 2003 13:06 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. Michael T Farnworth wrote: On Fri, 24 Oct 2003, Michael Koehler wrote: :) imo it is called Budge Tone .. budge from moveWhat will you guys think what the name HandyToneimply, which could suggest convenience ? Anything that gets rid of 'Budget' in the name is a good idea in my mind,as long as it isn't changed to CheapTone :-) 'Handy' is quite a goodchoice I think. If you tried to turn it into SuperTone it might be just viewed as being over the top and unbelievable, so you probably do need something in the middle. You could be right. DeskTone would be the right choise then maybe.. Michael Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
Re: [Asterisk-Users] Survey: Grandstream improvements.........
There's a fix I'd like : When you pick up the phone and press callers then send, any standard human being would like to have the number shown sent, not another one ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
:) imo it is called Budge Tone .. "budge" from "move" What will you guys think what the name "HandyTone"imply, which could suggest convenience ? The BT looks better as the majority of all hotel room phone i've ever seen in the US. Dave Weis wrote: On Wed, 22 Oct 2003, Michael T Farnworth wrote: It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. Second that. It does look/sound cheap.
Re: [Asterisk-Users] Survey: Grandstream improvements.........
At 09:15 PM 10/23/2003, you wrote: :) imo it is called Budge Tone .. budge from move What will you guys think what the name HandyToneimply, which could suggest convenience ? handi-tone (handicapped features) ?? The BT looks better as the majority of all hotel room phone i've ever seen in the US. Dave Weis wrote: On Wed, 22 Oct 2003, Michael T Farnworth wrote: It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. Second that. It does look/sound cheap.
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 2003-10-22 at 07:44, Andrew Kohlsmith wrote: Can you _please_ trim the quoted text? There's absolutely no reason to quote the entire post you're replying to, signature lines and all... +2 points for bottom-posting though. :-) No, -10 points for bottom-posting but not trimming. If you're not going to trim, I'd prefer you save me the hassle of scrolling and top-post. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 2003-10-21 at 17:13, John Brown (CV) wrote: Can you provide more specific information. Saying Its Broke Jim doesn't provide enough content :) True that. :) My biggest complaint was how they used to sometimes take over the server's MAC address, confusing the crap out of my switch. We only detected that because we were on an HP ProCurve that we could log into and view stats on, and the MAC address kept switching between two ports. But that is fixed in the .81 release, thankfully. However, it doesn't give me much faith in their TCP/IP stack... The switch they don't work with now is a CompUSA brand 8-port switch. I don't know the model number. I admit that it's a cheap switch, but it works with everything else in my house. With the BT phones plugged in, weird things happen. When I try to access the BT web page, the phone will give me the login page fine, but when I post the password, it freezes. As in, the phone requires a hard reset, it doesn't respond at all after 20 seconds or so. I tried to look at it in Ethereal, but everything seemed normal. I have no more data than that. I replaced the switch with a Linksys, and the phones no longer lock up now. What version of code are you running on the GS ?? 1.0.3.81 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes Here is another thought that I haven't heard mentioned... How about changing the TFTP upgrade in favour of HTTP upgrades and config file retrieval.. I am sure almost everyone has an HTTP server available to them but I doubt many have a TFTP server available.. I think this would help many people.. If you agree reply.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
WipeOut wrote: John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes Here is another thought that I haven't heard mentioned... How about changing the TFTP upgrade in favour of HTTP upgrades and config file retrieval.. I am sure almost everyone has an HTTP server available to them but I doubt many have a TFTP server available.. I think this would help many people.. If you agree reply.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Virtually every linux distribution I know of has TFTP as part of the distribution, or is easily available as an add on. It is trivial to set up, has very low overhead and a small footprint. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Stephen R. Besch wrote: WipeOut wrote: John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes Here is another thought that I haven't heard mentioned... How about changing the TFTP upgrade in favour of HTTP upgrades and config file retrieval.. I am sure almost everyone has an HTTP server available to them but I doubt many have a TFTP server available.. I think this would help many people.. If you agree reply.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Virtually every linux distribution I know of has TFTP as part of the distribution, or is easily available as an add on. It is trivial to set up, has very low overhead and a small footprint. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I still think HTTP is a better option.. There is far more control available in terms of securing it especially when the description of the package says TFTP provides very little security, and should not be enabled unless it is expressly needed... The discussion is really moot since I doubt they will change it anyway.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Can you _please_ trim the quoted text? There's absolutely no reason to quote the entire post you're replying to, signature lines and all... +2 points for bottom-posting though. :-) I still think HTTP is a better option.. There is far more control available in terms of securing it especially when the description of the package says TFTP provides very little security, and should not be enabled unless it is expressly needed... Who in their right mind is putting these phones on the open Internet, and if they're not, then why is TFTP that big a problem? TFTP's actually quite a standard option in most networking equipment for pulling down new configurations and firmware. HTTP doesn't offer much in the way of helping with that. What would be nice is perhaps a little DIP switch on the phone to enable LAN reconfigure/flash for better security... but for me anyway being able to pull the TFTP server and config filenames (global and per-phone) from standard DHCP extensions would be awesome. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
http == hyper text transport protocol tftp == trivial FILE trasfer protocol thus using tftp to do updates seems better. Its also a smaller foot print code wise and in boot loader thats important. tftp servers are available, On Wed, Oct 22, 2003 at 08:58:33AM +0100, WipeOut wrote: John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes Here is another thought that I haven't heard mentioned... How about changing the TFTP upgrade in favour of HTTP upgrades and config file retrieval.. I am sure almost everyone has an HTTP server available to them but I doubt many have a TFTP server available.. I think this would help many people.. If you agree reply.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, Oct 22, 2003 at 02:24:57PM +0100, WipeOut wrote: Here is another thought that I haven't heard mentioned... How about changing the TFTP upgrade in favour of HTTP upgrades and config file retrieval.. I am sure almost everyone has an HTTP server available to them but I doubt many have a TFTP server available.. I think this would help many people.. If you agree reply.. :) Virtually every linux distribution I know of has TFTP as part of the distribution, or is easily available as an add on. It is trivial to set up, has very low overhead and a small footprint. I still think HTTP is a better option.. There is far more control available in terms of securing it especially when the description of the package says TFTP provides very little security, and should not be enabled unless it is expressly needed... right, adding HTTPS and HTTP to the boot loader would cause that to inflate and possibly be to big to deal with. so enable tftp and put a couple of ipfw statements on the box to limit who can tftp from/to you. when tftp says it provides little security, that should really say tftp provides little to no authentication, ie it doesn't ask for a uid/pwd. http is a bad idea imho. I don't want to have to carry around a web server on my laptop, or have to have my customers config a web server to deal with updating their phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: http == hyper text transport protocol So are the entries on your hard drive with a .htm or .html extension not files? (sorry a little sarcastic I know) tftp == trivial FILE trasfer protocol thus using tftp to do updates seems better. Its also a smaller foot print code wise and in boot loader thats important. The boot loader size is the the best argument I have heard so far for using TFTP, but memory is pretty cheap now compared to the days gone by.. :) tftp servers are available, later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: right, adding HTTPS and HTTP to the boot loader would cause that to inflate and possibly be to big to deal with. True.. so enable tftp and put a couple of ipfw statements on the box to limit who can tftp from/to you. Could be made to work but most IP's are dynamic.. when tftp says it provides little security, that should really say tftp provides little to no authentication, ie it doesn't ask for a uid/pwd. Also true.. http is a bad idea imho. I don't want to have to carry around a web server on my laptop, or have to have my customers config a web server to deal with updating their phone. I would think setting up a web server would be easier than setting up a tftp server.. but thats just my opinion.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Andrew Kohlsmith wrote: Text trimmed.. :) Who in their right mind is putting these phones on the open Internet An example.. sipphone.com , and if they're not, then why is TFTP that big a problem? TFTP's actually quite a standard option in most networking equipment for pulling down new configurations and firmware. HTTP doesn't offer much in the way of helping with that. Its not as easy to upload to HTTP as it is to upload to TFTP.. :) What would be nice is perhaps a little DIP switch on the phone to enable LAN reconfigure/flash for better security... but for me anyway being able to pull the TFTP server and config filenames (global and per-phone) from standard DHCP extensions would be awesome. What is the phone is getting its IP from a DHCP server that you don't have control over??.. eg a home worker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, John Brown (CV) wrote: http == hyper text transport protocol tftp == trivial FILE trasfer protocol Based on this definition we could suggest that the web should only consist of a few html files as a jpeg clearly isn't hypertext. I suspect the reason HTTP was proposed is that almost everybody who runs a network will know about it, and we must also remember that almost every Unix OS is likely to come with a pre-installed and configured webserver, on the other hand tftp is almost always going to be disabled and need configuring. HTTP is trivial to implement on the client side, you simply send a line like: GET /someconfig.txt HTTP/1.0 You then read the input until you get to a blank line, then the file follows that. It isn't hard to write a simple http daemon if people are looking for a small footprint, in fact I am sure such things do exist. You could even write a http daemon using tcp_wrappers and a few lines of shell script if hard pressed. I suspect tftp probably has a simple protocol too. Maybe support could be added for http as well as tftp. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, Oct 22, 2003 at 03:15:27PM +0100, WipeOut wrote: http is a bad idea imho. I don't want to have to carry around a web server on my laptop, or have to have my customers config a web server to deal with updating their phone. I would think setting up a web server would be easier than setting up a tftp server.. but thats just my opinion.. setting up a tftp server on my laptop took less than 6 minutes, download program, install, point to directory containing files, point device to laptop's IP address ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, Michael T Farnworth wrote: I suspect tftp probably has a simple protocol too. Maybe support could be added for http as well as tftp. I take this back, as a protocol tftp is hideously complex compared to http and would take a lot more code. However tftp is based on tcp rather than udp so it requires less complex networking support. Amusingly though SIP has a lot in common with HTTP, so maybe half the work is done already, so much so that in one of the SIP RFC's they even go so far as to say ... Except for the above difference in character sets, much of the message syntax is and header fields are identical to HTTP/1.1; rather than repeating the syntax and semantics here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [11]). However they do also point out that: Unlike HTTP, SIP MAY use UDP. In fact I believe a SIP client doesn't have to support TCP, but fortunately I believe the Grandstream does. Feel free to point out errors in this brief bit of research. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Ouch.. you hit one of my pet peeves.. See below.. This is not meant to be a FLAME but rather SOAPBOX. Robert John Brown (CV) wrote: http == hyper text transport protocol So are the entries on your hard drive with a .htm or .html extension not files? (sorry a little sarcastic I know) *** Big difference beween httProtocol HyperTextMarkupLanguage :-) tftp == trivial FILE trasfer protocol thus using tftp to do updates seems better. Its also a smaller foot print code wise and in boot loader thats important. The boot loader size is the the best argument I have heard so far for using TFTP, but memory is pretty cheap now compared to the days gone by.. :) SOAPBOX Yes, memory is cheap, disk space is pratically free and processors increase in power every year. But that is not a reason to ignore memory usage or write inefficient programs. IF we used the same programming standards as we had in the last century :-) (70s and 80s) then WinXP would probably run on a 486 with 64MB RAM. /SOAPBOX From what I have seen, the Asterisk code must be fairly good. Its running quite nice on my P100, 32 MB system. MusicOnHold ran for 2,5 hours last night without any noticable distortion. Ok.. I don't have many phones hooked up but was fairly surprised that it does as well as it does. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
However tftp is based on tcp rather than udp so it requires less complex networking support. Most tftp implementations use udp (not tcp) with an added layer to identify missed or out of order packets. Slightly more complex then one would think, but not all that bad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 2003-10-22 at 15:38, Michael T Farnworth wrote: On Wed, 22 Oct 2003, Michael T Farnworth wrote: I take this back, as a protocol tftp is hideously complex compared to http and would take a lot more code. RFC 1350 (tftp v2): 11 pages RFC 2616 (http/1.1) : 114 pages josh. -- --- Josh Howlett, Networking Digital Communications, Information Systems Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On 22-10 15:38, Michael T Farnworth wrote: In fact I believe a SIP client doesn't have to support TCP, but fortunately I believe the Grandstream does. RFC3261 compliant SIP clients must support TCP. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, Michael T Farnworth wrote: However tftp is based on tcp rather than udp so it requires less complex networking support. Replying to own email here, which is bad I am told, but I did make a mistake, I meant to say tftp is based on udp rather than tcp. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On 22 Oct 2003, Josh Howlett wrote: On Wed, 2003-10-22 at 15:38, Michael T Farnworth wrote: On Wed, 22 Oct 2003, Michael T Farnworth wrote: I take this back, as a protocol tftp is hideously complex compared to http and would take a lot more code. RFC 1350 (tftp v2): 11 pages RFC 2616 (http/1.1) : 114 pages There might be lots more options for http (if you want to use them), but using http in its simplest form is easy. You can do a http request using telnet and typing a single line. I doubt that is possible with tftp (even if there was a udp equivalent of telnet). Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. The current name resulted in my wife bursting into laughter and saying 'I can't believe they called it that.' I suspect clients are more likely to say 'perhaps we need the professional product if we are going to risk our business on it.' Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 2003-10-22 at 18:50, Michael T Farnworth wrote: It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. Hadn't thought of that until you mentioned it. But it's perhaps because living in a non-English speaking country I see English words totally misused that I've come to accept it. For instance I just bought a case to build up a machine it's called an Enjoy? Why not just BT-101, BT-102, BT-102D? That's unless you're in the UK where BT could have the same implications as your original point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Can you _please_ trim the quoted text? There's absolutely no reason to quote the entire post you're replying to, signature lines and all... +2 points for bottom-posting though. :-) I still think HTTP is a better option.. There is far more control available in terms of securing it especially when the description of the package says TFTP provides very little security, and should not be enabled unless it is expressly needed... Who in their right mind is putting these phones on the open Internet, and if they're not, then why is TFTP that big a problem? TFTP's actually quite a standard option in most networking equipment for pulling down new configurations and firmware. HTTP doesn't offer much in the way of helping with that. What would be nice is perhaps a little DIP switch on the phone to enable LAN reconfigure/flash for better security... but for me anyway being able to pull the TFTP server and config filenames (global and per-phone) from standard DHCP extensions would be awesome. Regards, Andrew Anybody running an IPCSP is in their right mind, and that's completely over the open Internet. The argument here is not necessarily one of ease of use, but of security. John Brown said in a later post that TFTP's problem was not security, but authentication. Close, but not quite. The problem with TFTP is that it is neither authenticated NOR encryptable by nature. I have no issue with the lack of authentication if the files moved can be encrypted. This is a critically important point: sending out cleartext TFTP (or HTTP, for that matter) files across ANY network is ill-advised. Grandstream can stick with TFTP, or use HTTP, I don't care which. Anyone who has enough of these phones that they're dynamically re-configuring them with a file transfer mode will figure out one or the other. The issue is that there MUST (MUST, MUST, MUST) be a way to encrypt the files so that someone grabbing them from some non-GS client or intercepting the communications on the wire cannot get passwords or any useful data from the file. Otherwise, no IPCSP in their right mind would ever implement the Grandstream phones, ever, in any way, period. And, while enterprise users are a decent market share, I don't think Grandstream would say that IPCSPs (who will be ordering 1000's of these at a time) can be ignored. See the Cisco ATA-186 for a well thought-out implementation of this method using TFTP. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, John Todd wrote: The problem with TFTP is that it is neither authenticated NOR encryptable by nature. I have no issue with the lack of authentication if the files moved can be encrypted. This is a critically important point: sending out cleartext TFTP (or HTTP, for that matter) files across ANY network is ill-advised. I'll add one potential problem to TFTP in the current internet as we know it. With all the recent worms of this summer it has been many vendors recommendations to block tftp. I've seen increasing number of cable/dsl providers following these recomendations. So over the public internet software/config grabbing with tftp could be a potential problem James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 22 Oct 2003, Michael T Farnworth wrote: It just struck me that the easiest improvement would be to drop the name BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests poor quality. The phones need a name which implies 'High Quality'. Second that. It does look/sound cheap. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I got these suggestions from the voip forum on dslreports: 1- making them mount on the wall without the handset falling off. They have the screws, but no hook thing on the handset. 2- why do you have to lift the handset to see who called? being able to scroll thru the call display without lifting the handset would be cool 3- fix bugs 4- more codecs. g.711 is the only codec i can use with asterisk which is kind of annoying because it chews through a lot of bandwidth. gsm or ilbc would be really nice. they are free and use very little bandwidth and have releatively decent call quality (at least if you're used to using a cell phone a lot) -Mark --- gsm or ilbc would be really nice. --- Boost audio level in speaker phone mode. The speaker phone audio level is very marginal... The room have to be very quiet and you better be no more than two listening as you will feel crowded getting close enough to the phone to be able to hear. Alain - How about improvements to the design. 1) Music on hold. Add a memory slot to the side of the phone so you can load in a few songs that will play when someone is put on hold. Or somehow be able to load songs into the phone through the web browser interface. 2) A second line that allows you to make and receive PSTN calls with an extra feature for conferencing between the two lines. I like to only have one phone on my desk that can be used for both PSTN and IP calls. All be it, I do have a two line analog phone and am waiting for the ATA-286 which will solve this problem. 3)Speed dial keys for the most dialed numbers. 4)Headset jack. 5)Different Ring Tones. 6)Louder Speaker volume. Thanks for listening. Martin - One that would be particularly useful since most VOIP providers don't offer 911 is Fire/Ambulance/Police speed-dial buttons, and an optional 911 override in the firmware so if someone tries to dial 911 it can tell them to press one of the programmed buttons- if they're not programmed advises them to find an alternate phone. Make the 911 override optional as some VOIP providers and/or local soft PBX's may have 911 service (I have 911 programmed into asterisk to present a menu of 1 for police 2 for ambulance 3 for fire) and routes the calls according to the context they are in- since my physical location (and that of other friends/employees who have btones) is different from where my actual PSTN connectivity is. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc John - I've discussed some of these with you, but here are a few for consumption and comments by the list: 8 - Encryption. * will probably support it as soon as some reasonable handsets support it. Grandstream should be the initiator in this process, with SRTP or some other RFC-approved method for delivering crypto SIP audio channels. Every business I talk to lists this on their priority chart for VoIP, and there are NO ANSWERS right now from major SIP handset vendors as far as voice crypto goes. I'm starting to think that it's a conspiracy. 8 - The TFTP configuration nonsense has been discussed. This needs to turn into an openly documented standard. Proprietary standards are useless, and all of them die eventually - why prolong the agony on your customers? 5 - Weight. Phone should weigh more. I'm constantly pulling it across the table with only the slightest stretching of the phone cord. 6 - Tilting display. Display should tilt up so I can actually read it. OR: base that tilts the whole phone up about 45 degrees (note: if this is the case, the weight issue really needs to be resolved) 9 - Buttons. The 102 model I have absolutely SUCKS as far as the buttons go. I have to pretty much press them like manual typewriter keys to get them to work. Any lateral force causes them to bind up. 10 - button response. Even when I _do_ manage to press the keys firmly enough, if I type too fast the keystrokes are lost. This is really, REALLY annoying. Button response needs to be sped up significantly. I almost always have to dial every number two or three times, or slow down to one button every second. Thus, I use my Cisco phones and leave the grandstream to gather dust. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. 1. More volume out of the speakerphone, and better range of the headset volume. I guess it would be sort of out there but if it were possible to separately adjust them that would be boss. To get the speakerphone to even be heard whilst hunching over it requires full volume. But then if someone calls and I don't put it back down it blasts my ears off. 2. Support for lower-b/w codecs. My list would include iLBC, Speex, and GSM. 3. Announced (supervised? consultative?) Transfers. 4. IAX support, which would lead to better NAT support. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes Hi John, My biggest issue is a hardware issue and is the single biggest reason why I have not been able to sell the GS phones into a company and that is the 10Mbps ethernet ports.. I guess if you are using the 101 then its not much of an issue but the whole cost saving is to cut down on wiring costs so the only model we even look at it the 102.. I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Staying with hardware, the screen needs to be angled a little to make it easier to read and needs to support more digits, and the buttons need to be easier to press.. Here are my suggestions for firmware updates.. 10 - Support for open low bandwidth codecs, specifically iLBC and GSM. 10 - Consultative Transfer. 7 - A nice feature of the Snom phones is the ability to type in the number with the handset still down and then the number is dialed when the handset is lifted or the OK button is pressed. This way you can take as long as you like to dial a number.. GS have the send/dial button so this feature should not be hard to add.. Adding to this.. it would be nice to be able to go through the Called and Callers call logs with the handset down and then when on the number you want to dial just lift the handset.. 5- Config refresh, apply config settings (even some of then) without needing to reboot the phone. Mark on the config page which settings will require a reboot to take effect.. 3 - Show the text part of the CallerID..(Think this may be a hardware issue or limitation) Fianally hardware support.. I had a power supply go on one of my GS phones, I purchased that phone from GS's agent in the US before there was anywhere to buy the phones in the UK, I contacted GS and then appologised and asked for my address, I assumed that was so I could be sent a replacement, I sent them my address..Now months later I have sent follow up emails which never get a response and I still don't have a replacement power supply.. so maybe you can speak to the president about that too.. That should about do it.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 20 Oct 2003, John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. It goes without saying that consultative transfer has to be a 10 and I am sure I am not alone in saying so. Other things are niceties, but when selling to business this is an expected basic minimum. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 20 Oct 2003, John Todd wrote: 9 - Buttons. The 102 model I have absolutely SUCKS as far as the buttons go. I have to pretty much press them like manual typewriter keys to get them to work. Any lateral force causes them to bind up. 10 - button response. Even when I _do_ manage to press the keys firmly enough, if I type too fast the keystrokes are lost. This is really, REALLY annoying. Button response needs to be sped up significantly. I almost always have to dial every number two or three times, or slow down to one button every second. Thus, I use my Cisco phones and leave the grandstream to gather dust. I found that the buttons didn't work very well and I had lots of repeated or missed digits, making it almost impossible to login to the voicemail. However when I moved from using RTP to SIP INFO the problem vanished. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, rnc Info Lists wrote: 9 - ability to switch back and forth between speakerphone and handset The Grandstream seems to have a strange method of working when it comes to speakerphone. I would expect the speakerphone button to just switch on and off the speaker, however it doesn't. If during a call you switch on the speaker then if you press th speakerphone button again to switch it off it hangs up the phone. However if you put the phone down instead and then pick it up again the speaker goes off and the call remains connected. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 20 Oct 2003, John Todd wrote: 9 - Buttons. The 102 model I have absolutely SUCKS as far as the buttons go. I have to pretty much press them like manual typewriter keys to get them to work. Any lateral force causes them to bind up. 10 - button response. Even when I _do_ manage to press the keys firmly enough, if I type too fast the keystrokes are lost. This is really, REALLY annoying. Button response needs to be sped up significantly. I almost always have to dial every number two or three times, or slow down to one button every second. Thus, I use my Cisco phones and leave the grandstream to gather dust. I found that the buttons didn't work very well and I had lots of repeated or missed digits, making it almost impossible to login to the voicemail. However when I moved from using RTP to SIP INFO the problem vanished. Michael My issue is not the encoding of the digits into the data stream, but the ability of the device to recognize the keystrokes. I use INFO, as well, after the usual failed experiments with inband and RFC2833 encoding. It just seems like there is some hardware issue that is not fast enough to catch my key presses. This is even before the call is handed off to the proxy (initial dial) so it's not a data transfer problem... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Hey WipeOut, Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, Low, Adam wrote: Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? The cable goes into the phone and then out of the phone into the computer. That switch in the phone is 10Mbit so the computer ends up on 10Mbit too. Perhaps the best way to avoid this is to join all the phones together since they are all 10Mbit anyway, so you will then just need one extra ethernet socket in the room for all the telephones. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Bah, I replied directly instead of to the list. :-( 1 = Nice to have some day 10 = Got to have it right now 10 - Fix SIP disconnection problem 9 - Ringtones (downloadable?) 8 - ILBC 8 - MUCH MORE professional looking case (this includes dropping the four red LEDs beneath the white plastic face), maybe a nice black/gray/smoke matte plastic case, a wall mounting kit with a catch for the handset, etc. 7 - assisted transfer (I think that's what it's called) 6 - POE (12V-48V input range) 6 - 2.5mm headset jack 5 - integrated 100mbit switch ***capable of sustaining 100mbit*** 3 - IAX/IAX2 would be VERY nice 1 - downloadable codecs How's that for starters? Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, Low, Adam wrote: Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? The cable goes into the phone and then out of the phone into the computer. That switch in the phone is 10Mbit so the computer ends up on 10Mbit too. Perhaps the best way to avoid this is to join all the phones together since they are all 10Mbit anyway, so you will then just need one extra ethernet socket in the room for all the telephones. Michael Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 21 Oct 2003, rnc Info Lists wrote: Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. On the subject of collisions it seems to me that individual phone bandwidth use is relatively limited when compared to the 10Mbit/s available, so would the problem really be that substantial? Personally I currently have: Hub - Phone - Phone - Laptop No visible problems here, so certainly 3 phones in a line would seem to work. I suppose it all comes down to how many phones you put in a line. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
6 - 2.5mm headset jack 6.5 - when a headset is connected the ringer should NOT come through the headset... damn that is annoying on softphones... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On 21/10/2003 11:14, Andrew Kohlsmith wrote: [...] 6 - POE (12V-48V input range) [...] 5 - integrated 100mbit switch ***capable of sustaining 100mbit*** [...] +1 on both of these points. The power brick is cheap and nasty. POE would be a huge plus. A 100mb bridge would make the phone a lot more attractive in an office full of cables. I'd also add my voice to the request for a better speakerphone. The dialtone comes out loud and clear but everything else is too muted. If I up the volume to hear calls, then the dialtone becomes deafening - as does the handset when used. I'm less concerned about the codecs as I'm happy to use ULAW/ALAW on the internal network and have Asterisk transcode to something else for external calls. There should be a way of locking the menu button, as it is too easy to muck with the settings. For central configuration, the cfg.txt file format would be nice, but is still a pain. Ideally I'd like to be able to configure the phone via DHCP extensions. That would be ideal as I can configure the lease time to manage how frequently the phones update and I can centralise the configuration with the IP details. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
+1 on both of these points. The power brick is cheap and nasty. POE would be a huge plus. A 100mb bridge would make the phone a lot more attractive in an office full of cables. I specifically stated a wide POE range because let's face it, with the power requirements that phone has, a wide-input-range DC-DC converter is _peanuts_, especially if you've already got a tiny switchmode converter for line power. A very wide range on POE input makes it easy to mix and match phones too. Hell if you've got a switcher already, you can make it autosense polarity too. Don't pull a Cisco. Don't try and lock your users in to one brand of switches. As for the 100mbit switch -- again I was very specific here -- don't throw on one of those $0.25 100 mbit switch chips that can only sustain about 1MB/sec -- I put in a 100mbit switched network to achieve 11MB/sec sustained, not burst. A two-port switch capable of full sustained network speed shouldn't be expensive and can really be a big marketing feature. We won't screw your network speeds kind of thing. :-) I'd also add my voice to the request for a better speakerphone. The dialtone comes out loud and clear but everything else is too muted. If I up the volume to hear calls, then the dialtone becomes deafening - as does the handset when used. Speakerphone is a big deal with me too. For central configuration, the cfg.txt file format would be nice, but is still a pain. Ideally I'd like to be able to configure the phone via DHCP extensions. That would be ideal as I can configure the lease time to manage how frequently the phones update and I can centralise the configuration with the IP details. Why not specify a TFTP server/config filename via DHCP? It's already standard and would work very well. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I love to have on my GS, GSM codec, scale = 10 - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 21, 2003 1:48 AM Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. 7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Andrew Kohlsmith wrote: Why not specify a TFTP server/config filename via DHCP? It's already standard and would work very well. This would need to be optional, what if a phone was deployed remotely where you have no control over the DHCP.. then you would need to specify the config file location or statically set the config.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
Hi! I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. Correct me if I am wrong: 5 switches on 10 Mbit/s 2 switches on 100 Mbit/s (for the same segment) Note: Switches slow down your network... cable length matters as well, of course. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Philipp von Klitzing wrote: Hi! Correct me if I am wrong: 5 switches on 10 Mbit/s 2 switches on 100 Mbit/s (for the same segment) Note: Switches slow down your network... cable length matters as well, of course. Philipp IIRC it was 5 HUB's on 10Mbps and 2 HUB's on 100Mbps, I seem to rememeber that when switches came along the rules got trashed and each manufacturer made their own rules.. But I could be wrong.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
It goes without saying that consultative transfer has to be a 10 and I am sure I am not alone in saying so. Other things are niceties, but when selling to business this is an expected basic minimum. I fully agree with that. on my list, 'supervised transfer' is the more (software) feature needed. then goes ringtones and at least gsm codec On the hardware point of view : real 100Mbits interface, heavier case ? matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John, I second Brian's comments. After setting up 20 GS phones using their somewhat odd web interface, I would really appreciate a more rational provisioning system for small to medium installations. I would add the following: cfgEveryone.txt:Generic setup for all phones. Read first - and overridden by cfgMACADDRESS.txt:the specific setup items for each phone. Note, that there already seems to be a config file format (undocumented). If this is true, GS should at least publish the format and let the OS community have a go at a configurator. Also, the daylight savings option will ultimately need to be fixed to include date recognition. The current setup requires that you log into every phone twice each year to turn the option on/off. For installations with a large number of phones, this is going to be a real headache. And, the speakerphone button needs to be fixed. It works now almost perfectly. The only glitch is when you are on the speakerphone and want to switch back to the handset. If the handset is on the cradle, picking it up will transfer the call to the headset from the speakerphone. However, if you have the handset off hook already and press the speakerphone button expecting to transfer back to the handset, you are disconnected. The documentation states that it is a toggle. It isn't. The workaround is to press the on-hook button momentarily and you are switched back to the handset. Nevertheless, the speakerphone button should not hang up the line unless the receiver is already on hook. Finally, the documentation for IP QOS, VLAN Tag and Dialplan need to be expanded/included. Stephen R. Besch John, I want the tftp configs done like cfgMACADDRESS.txt or compile them into a binary form like the ATA's use. And stop trying to rip us for the GAPS system. WHAT A RIP. It makes cisco so worth the extra cash! Config refresh similar to the ATA.. refresh config every x seconds. bkw On Mon, 20 Oct 2003, John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
I have a Nortel phone on my desk right now. IF the handset is picked up and you press the speaker button, it does not hang up but switches back to the handset instead. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Hajime Lanning Sent: Tuesday, October 21, 2003 10:26 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. quote who=Michael T Farnworth On Tue, 21 Oct 2003, rnc Info Lists wrote: 9 - ability to switch back and forth between speakerphone and handset The Grandstream seems to have a strange method of working when it comes to speakerphone. I would expect the speakerphone button to just switch on and off the speaker, however it doesn't. If during a call you switch on the speaker then if you press th speakerphone button again to switch it off it hangs up the phone. However if you put the phone down instead and then pick it up again the speaker goes off and the call remains connected. I never had this problem. As all the PBX phones (currently NorTel Meridian) that I have used work that way. (Speaker button turns on the speaker, use hook button to switch back to handset.) -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
(Ethernet issues) RE: [Asterisk-Users] Survey: Grandstream improvements.........
quote who=Michael T Farnworth On Tue, 21 Oct 2003, rnc Info Lists wrote: Michael, How would you be able to connect all phones in a room to one socket? The Ethernet specificiation has a limit to the number of hubs/switches that can be inline. (or at least it used to). The only way I can see to connect all phones to one socket would be to daisy chain them. This would not be a good solution since: - all phones would use the same 10mbps segment, chances for collisions would be high - rules of Ethernet would be violated so even if it did work it may stop at any point with some other normally minor change. I defer to your knowledge in this area, but I would be interested to know what the limit is in terms of the number of devices that can be put inline. On the subject of collisions it seems to me that individual phone bandwidth use is relatively limited when compared to the 10Mbit/s available, so would the problem really be that substantial? Personally I currently have: Hub - Phone - Phone - Laptop No visible problems here, so certainly 3 phones in a line would seem to work. I suppose it all comes down to how many phones you put in a line. Michael Too many switches/hubs will cause late collisions. Late collisions are ethernet collisions that happen after the transmitting station has finished transmitting. If it is a store and forward switch, then the switch can retransmit on collision, otherwise the packet is completely lost. This is the same reason why an ethernet cable cannot be over 300 feet. The first bit of the ethernet frame must get to the farthest node in an ethernet segment before the last bit is transmitted by the originating station. This length is based on speed one bit takes to span the distance and the minimum ethernet frame size (64 bytes). Currently the limit is 5 non-store and forward switches/hubs. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I never had this problem. As all the PBX phones (currently NorTel Meridian) that I have used work that way. (Speaker button turns on the speaker, use hook button to switch back to handset.) Agreed. One thing that would be nice though is to emulate the meridian's use of the handsfree button as a mic mute toggle when in handsfree mode. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I have a Nortel phone on my desk right now. IF the handset is picked up and you press the speaker button, it does not hang up but switches back to the handset instead. Not with my Meridian system. Just tested to verify: handset onhook + handsfree/mute pressed: handsfree (goes off-hook) handset offhook + handsfree/mute pressed: handsfree handsfree offhook + handsfree/mute pressed: mic mute toggle In all cases, to get back to handset use you must toggle hook switch. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tuesday 21 October 2003 01:07, John Todd wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. 5 - Weight. Phone should weigh more. I'm constantly pulling it across the table with only the slightest stretching of the phone cord. I'd have to respectfully disagree. If this is really a problem I'd suggest taking advantage of the mounting bracket on the bottom and either attach the phone to the desk or attach a sheet of lead. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: (Ethernet issues) RE: [Asterisk-Users] Survey: Grandstream improvements.........
Personally, I just wire every jack the same way back to the patch panels, 4pr cat5 or better, terminated in an rj45. Back at the panel wire the blue pair to your analog telephony stuff, and the org/grn to your networking. Then if you plug in an rj11 you get a phone line, if you plug in a network cable that works too. Some would say this is wasteful of wire, but in reality the wire is the least part of the cost of a cabling installation. Labour far outweighs it. If you want a physical 10mb/sec subnet for your phones, easy, just patch the relevant jacks into that hub/switch, separated from the jacks used from your data network. There are also some ways to stretch this distance limit if you are careful, and limit the branching topology of the lan segment. This is the same reason why an ethernet cable cannot be over 300 feet. The first bit of the ethernet frame must get to the farthest node in an ethernet segment before the last bit is transmitted by the originating station. This length is based on speed one bit takes to span the distance and the minimum ethernet frame size (64 bytes). Currently the limit is 5 non-store and forward switches/hubs. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
We have a 10 and we need it yesterday (as well as many other people who don't even know it). We have a Bug report at GS. The problem is with STUN and changing IP Addresses. It happens like this: 1. Phone does a STUN query and registers fine. 2. If the public IP Address changes sometime later (like on a DSL line that disconnects and connects back), the phone will keep registering with the original IP address, and thus will fail to work properly. It apparently does not attempt further STUN queries for registration purposes. STUN isn't even needed nat=yes is all you need and it just works. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
My issue is not the encoding of the digits into the data stream, but the ability of the device to recognize the keystrokes. I use INFO, as well, after the usual failed experiments with inband and RFC2833 encoding. It just seems like there is some hardware issue that is not fast enough to catch my key presses. This is even before the call is handed off to the proxy (initial dial) so it's not a data transfer problem... I use RFC2833 and it works fine... as for switching to and from handset and speakerphone it can be done. press speaker phone ... your on speaker phone... hangup the handset.. then when you pick the handset back up you are on speaker phone. just an FYI bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
I alwasy laff at those DISCLAIMERS on email... funny they are at the bottom. bkw On Tue, 21 Oct 2003, Low, Adam wrote: I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Hey WipeOut, Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10 Fix call waiting tone. 9Fix the tftp configs so that I can host my own provisioning server. Or make a command prompt based tool kit, so that I can use Gaps with out writing a http screen scraper. 4 Having the Conference button do something would be cool. John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tuesday 21 October 2003 10:52, Brian West wrote: We have a 10 and we need it yesterday (as well as many other people who don't even know it). We have a Bug report at GS. The problem is with STUN and changing IP Addresses. It happens like this: 1. Phone does a STUN query and registers fine. 2. If the public IP Address changes sometime later (like on a DSL line that disconnects and connects back), the phone will keep registering with the original IP address, and thus will fail to work properly. It apparently does not attempt further STUN queries for registration purposes. STUN isn't even needed nat=yes is all you need and it just works. We only use Asterisk for PSTN calls. All our subs register in SER, and sure, we could also do the above trick in SER as well, but that would force the RTP stream to pass though our server. We try to avoid it if possible and STUN is a great way to do it. Regards, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10. Auto answer option on 2nd line appearance. To support paging over the phones. Lee - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 10:38 PM Subject: [Asterisk-Users] Survey: Grandstream improvements. Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10. Auto answer option on 2nd line appearance. To support paging over the phones. That would be very cool. Voice Call I think it's called on the Meridian system. DND would be nice too (just return busy) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 2003-10-21 at 11:36, James Sizemore wrote: 9Fix the tftp configs so that I can host my own provisioning server. Or make a command prompt based tool kit, so that I can use Gaps with out writing a http screen scraper. So I'm not the only one who wrote an http screen scraper to handle configuring a network of phones? :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote: So please rate your ideas on a scale of 1-10 10 - Fix the TCP/IP stack. The phones don't work with certain switches (i.e. the one at my house), and occasionally do other weird things (although they fixed the MAC address takeover bug, apparently). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Agreed, don't drive up my shipping cost. light is good. Tilghman Lesher wrote: I'd have to respectfully disagree. If this is really a problem I'd suggest taking advantage of the mounting bracket on the bottom and either attach the phone to the desk or attach a sheet of lead. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
This is just my 0.02 but I would really like to have a headset jack. It means all the world to me, but I don't know about others. This would be extra important for cheapo call center clients (like I want to do). This would probably make the difference for me to not get a GS and I decide to buy a hardphone. I can't really put a fair number on this one. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10 - Alphanumeric Display. There is nothing more important. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Kevin Bockman wrote: This is just my 0.02 but I would really like to have a headset jack. It means all the world to me, Me too. I might buy a new GS if it has one :) -- JustThe.net Internet Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Can you provide more specific information. Saying Its Broke Jim doesn't provide enough content :) What version of code are you running on the GS ?? On Tue, Oct 21, 2003 at 01:33:48PM -0600, Steve Meyers wrote: On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote: So please rate your ideas on a scale of 1-10 10 - Fix the TCP/IP stack. The phones don't work with certain switches (i.e. the one at my house), and occasionally do other weird things (although they fixed the MAC address takeover bug, apparently). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
John Brown (CV) wrote: I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Hi John, Here are my suggestions for firmware updates.. 10 - Support for open low bandwidth codecs, specifically GSM. 10 - Support for working behind a NAT (same as SNOM). 10 - Open TFTP for firmware upgrades. 10 - let's skip anything else below 10. That would be a good start ... Uriel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
5 - ringer volume 7 - 'message' button should flash for message waiting rather than LCD 5 - LCD backlight can be set to always on 3 - wall mount hook for handset 8 - ability to lock the menu on the phone to stop users from 'playing' 7 - speakerphone is not loud enough, even when turned up full 10 - Announced / supervised / consultative Transfers 8 - 100mb ports rather than current 10mb (espec on pass-through models) 5 - angled LCD screen (adjustable would be great!) 7 - alphanumeric LCD 5 - get rid of the 4 red led's under the keypad (ugly!!) - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 21, 2003 3:38 PM Subject: [Asterisk-Users] Survey: Grandstream improvements. Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
John, I want the tftp configs done like cfgMACADDRESS.txt or compile them into a binary form like the ATA's use. And stop trying to rip us for the GAPS system. WHAT A RIP. It makes cisco so worth the extra cash! Config refresh similar to the ATA.. refresh config every x seconds. bkw On Mon, 20 Oct 2003, John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
I'm pretty happy with mine, I've got 2 of them as basic extensions, but I've found the following with daily use. The phone needs more lower bandwidth codecs, starting with GSM or ilbc scale 10 The blue backlight to stay on since the display doesn't tilt it makes it easier to see. flashing it for message waiting indicator. scale 10 The format for the tftp needs to be disclosed, GAPS or whatever they call it seems sort of we sold you a cheap phone, now we want to gouge you on the support for it and that doesn't make anyone happy. scale 10 Unrelated to firmware... the next models need to have wall hooks in the receiver and cradle for wall mounting. I'm sure I'll think of more things after I hit send, Mark (Q-At-Work in IRC) At 08:38 PM 10/20/2003, you wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Monday 20 October 2003 21:38, John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Hi John, We have a 10 and we need it yesterday (as well as many other people who don't even know it). We have a Bug report at GS. The problem is with STUN and changing IP Addresses. It happens like this: 1. Phone does a STUN query and registers fine. 2. If the public IP Address changes sometime later (like on a DSL line that disconnects and connects back), the phone will keep registering with the original IP address, and thus will fail to work properly. It apparently does not attempt further STUN queries for registration purposes. We have sent Sniffer traces to David Li and hopefully they will fix it soon. But we reported this over 4 weeks ago and it is still a huge problem for us. Please help us push this through. Thanks, Andres Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Survey: Grandstream improvements.........
10 - A way to lock the phone settings (IP address, etc). It is too easy to change the settings when in a public environment. The MENU button should not be 1 press away from changing the settings, Use MENU + SOME COMBINATION. 7 - Use the conference button to access Meetme. Like the Voice Mail UserID and Offhook Auto-Dial where you can preset an extension. OR call the Button Conference/Queue. 8 - Crank up the speakerphone volume. In a public place with background noise it is too soft. 8 - Have a model with a PSTN jack. There is a break out notch so that the phone can be used as a regular analog phone. Some H323 phones have this and it is very handy. 8 - Use better quality mouth pick transducers. The one used are too sensitive and clipping is noticeable. 9 - Mentioned before: The display is difficult to see, leave the back light on OR better still tilt the display up. My 2c contribution. MA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: Tuesday, 21 October 2003 2:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Survey: Grandstream improvements. 7 - Ringer volume control 4 - plug in module of user programmable buttons for frequently called numbers. Not everyone would need this so being able to add as an optional module would keep the base phone cost effective. 9 - ability to switch back and forth between speakerphone and handset 7 - message waiting light under the message button. The LCD light blinking is nice but is not easy to see when the room is well lit. 4 - headset jack Thanks for taking the survey. You might also encourage David to have his folks actively participate in the lists. I mentioned it to him before and his reason for not having a more active presence was to avoid the appearance of being commercial on the lists. Personally, I think that it would help to build a better relationship between his technical folks and their userbase. Robert Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
