similar to: Grandstream phones losing registration with server.

On Thursday 16 Jun 2005 09:25, Mark Brown wrote: > 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

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."

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

I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone

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 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

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, I have just enabled call parking on my Asterisk system and it's working well. However, I am running some Grandstream phones on my system and when you press # on them to transfer the call, the user on the other end of the line hears the tone produced by the phone, rather then Asterisk recognising that it's a command. Has anyone else has this problem and if so how did you correct it ?

Does anyone know of any reliable supplier for Grandstream phones? I tried dealing with David Li from Grandstream, but after emailing him an order in August, and asking how he wanted payment, I never got a reply... James Ho from DGTimes was happy to give me pricing, but when I sent him an email asking for shipping costs, I never got a reply... I tried dealing with John from Chagres Ventures, but

Hello, Right now I have app queue working with incominglimit=1, there is no call waiting signal, but after a while( like couple of hours) some phones randomly get stuck. The * thinks that they are in use and doesnt ring them, when they are infact not in use. sip show inuse, shows that they are inuse. typing reload on the console resets this and they are again available for working. anybody

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.

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them 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 ?

Just in case you guys haven't been paying attention Grandstream sliped in some diffrent colors on the IP phones and looks like they released the ATA-286 (Cisco is gonna have kittens I suspect) bkw

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

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:

For all thos Asterisk users not on the FWD list, it works for me!: -----Original Message----- From: Free World Dialup - The Future of Dialing [mailto:FWD@LISTSERV.PULVER.COM] On Behalf Of Leonidas Piagkos Sent: donderdag 12 juni 2003 0:58 To: FWD@LISTSERV.PULVER.COM Subject: Re: [FWD] FWD losing Registration Hi Don, All you have to do with your ATA is to set the following parameters as :

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

Anyone with any comments on DSS buttons and general phone features? Thanks, Shane -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050525/e11c72b7/attachment.htm

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully