similar to: Which GS IP product to buy??

There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and Dial() settings (Ttm). I tried all of the firmware on their BETA

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.

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Hi, I have used the Budgetone 102's extensively on Asterisk and found then quite reliable as long as you update the firmware. The GXP2000 is quite a mess at the moment as the current firmware does not support 3 quarters of the advertised functions and codec support is extremely limited. I have tested the unreleased latest firmware update for the GXP2000 and it's an incredible difference

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

hi .. i've just converted myself back to a newbie by trying to experiment with some new stuff .. I have connected two grandstream Budgettone 100 phones to my asterisk, and trying to experiment with them .. I am trying to get into the asterisk sample basically .. when I dial 1000 asterisk receives the call, but I do not hear any sound on the phone. Dialling from phone to phone direct (via

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

--- "Chandra" <chandra@digital.com.np> wrote: >i have been using grandstream budgettone IP phones and they work fine >except that these phones times out after some hours.. i ahve seen that >the phones working ok are next day unregistered and sip show peers do >not show their IP and although these phones can make calls , they >cannot be called. They Sip show peers

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

Hi, Yesterday I finaly got my handytone sip adaptor. It works.... But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry

I have used the handytone 488 from grandstream in the past.... However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. Jerry

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that asterisk@home has this built in to just a button hit, but i dont want to

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't