similar to: Grandstream phones losing registration withserver.

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

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc.

Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register.... Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '****' in the the two password fields... This is what caught me - I

Just got my (10) BT102 phones, flashed them to 1.0.5.20 and all work. No duds at all. Not a bad little phone at all. Doug

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actual measured at 8.86 mbps. The

Hi- I'm at my wit's end trying to get a Grandstream BT-101 POS to register on my * server. Asterisk version is 1.2.1. GS Firmware is rev 1.0.6.7. Basically, I've setup the phone following the instructions at voip-info.org, and it registers for about 10 seconds, then after receiving the SIP NOTIFY from the * server, goes into "flashing display" mode, which indicates some

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

Hi all, I'm quite new on this mailing list, and I discover the asterisk world. I m experimenting a PBX with SIP phones, grandstream budgetone (not expensive for tests) All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Here is my config : 2 sip phones BT102 with

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Would anyone have a copy of the latest firmware release for the grandstream BT102 phone? seems grandstream no longer offers it on their website (of if I missed something a link would be much appreciated.) Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090504/9ba2730e/attachment.htm

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.

Is there a a consensus on which of these is the better phone.I've personally been using an 841 and have learned to live with its shortcomings. I now need to recommend some phones for some sites we're installing. I'm looking at the BT102 for desktops that don't want/need a headset but need a phone for the higher end users (without costing the earth). TIA, tony Zero Effort

I was hoping someone can help with a problem with my GrandStream Budgetone "hanging" after a while. Problem seems related to the SIP registration - I am using dynamic registration (host=dynamic in sip.conf). Static IP is not an option in my case. I start Asterisk and all is groovy, phone works fine and can dial around. Then, at the time specified on the phone in the "Register

Hi, after upgrading to Asterisk 1.2.8 from 1.2.7.1 I got a problem with Grandstream BT100 after making an attended transfer (FLASH + NUMBER + SEND + WAIT ANSWER + TRANSFER). After the transfer, the display clears all the info except the clock, there is no dial tone, the WEB admin stops working. Incoming calls make the display light turn on but there is no ring and no callerid on the

Hi all, Im having problems using a fax machine conected trough a grandstream 286 sip ATA, it must be able to send and recive fax from pstn, but fax always ends with communication errors 252/244/232 and others. Im using alaw/ulaw codes on pass trough mode, also have tried asterisk faxdetection, nvfaxdetect, disable echo cancellation by hand always with same results. Grandstream ATA is using

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

The Grandstream 286s automatically re register when a connection is restored. Our Grandstream 2010s don't. Does anyone know of a setting that makes them reregister? I has tweaked "Watchdog timer" and anything that looked promising. Cary Fitch Affordable Telecom

I used to run Asterisk 1.4.4 but had to revert back to 1.2.13 to minimize a bug we were coming across. 1.4.5 looked promising, but the hints are broken and making it so I'll likely have to go back to 1.2.13 until I get the hints fixed. I'm using Grandstream phones & hints on the parked extensions. I should also clarify that when I upgraded versions, I renamed all Asterisk folders