similar to: Purchasing Grandstream Phones

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi, I've had a bunch of people asking about purchasing GS phones. I'm happy to announce that we now have an online purchase system available. We have: BT-101 BT-102 HT-286 Plus the full line of Digium hardware. We will be adding other products in the near future so check back :) Buyers are responsible for s/h costs. The URL for the phones is:

hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind

As I've said several times on this list[insert usual apology here], I still haven't received the last 20 of 100 phones I ordered over 2 months ago. If you get a hold of them please let me know MATT--- -----Original Message----- From: mikeu [mailto:mikeu@execpc.com] Sent: Saturday, January 10, 2004 12:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Chagres

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Hi, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 5. Grandstream hangs

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

Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will "glitch" or stutter randomly from time to time. Now, what is interesting is that this happens even with the phone totally disconnected from any

Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of "power hubs", but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be

Anyone have a good source for BT-101 phones? I had a lead on some, but they've not materialized. I'm also interested in the ATA-286 (HandyTone) units as well. This is for my personal Asterisk/INOC-DBA setup, that has yet to materialize heh. --- Tom Sparks

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit

Hi all, Just a little note for the records and archives. We see many small glitches / troubles in the mailing-list but rarely success stories ... Here's one : Asterisk is running perfectly fine in our setup : Debian 3.0 stable / Athlon 1.8, 256 MB Ram / Digium E-100P / Swisscom PRI isdn We have 6 companies (more to come) sharing the system in the building with a total of about 20

Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ

Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

Hi Everyone, I'm using Asterisk, actually A@H 1.1 with all Grandstream 102 phones. NAT is not an issue as all including the server have public IP's The problem is that the phones keep losing registration with the server. I have not timed this exactly to see if they drop off with exactly the same frequency. The SIP TRUNK connection to my provider SIPGATE does not lose registration, and

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at

Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to

I am using a grandstream phone with g729 and alaw odecs and in both modes I cannot seem to pass dtmf's, neither inband nor out of band, neither wthrough a lcoal server nor through a natted connection. Am I missing something ?